Abstract: Microphone arrays comprise several microphone capsules, the outputs of which being electronically combined for directional recording of sound. The directional and frequency properties of the microphone array depend on the number and positions of the microphone array. In order to obtain the smallest possible microphone array with only few microphone capsules, which, however, has an essentially uniform directional and frequency dependence over a speech frequency range, is scalable and robust against small incorrect positioning of the capsules, fifteen or twenty-one microphone capsules (K15,11 - K15,35, K21,11 - K21,37) are arranged on a carrier such that they lie on three similar branches, each with the same number of microphone capsules, which are rotated against each other by 120°. Each of the microphone capsules lies on a corner of a triangle of a grid in a flat isometric coordinate system with three axes rotated by 60° against each other and forming the grid_of equilateral triangles.

Abstract: Microphone arrays comprise several microphone capsules, the outputs of which being electronically combined for directional recording of sound. The directional and frequency properties of the microphone array depend on the number and positions of the microphone array. In order to obtain the smallest possible microphone array with only few microphone capsules, which, however, has an essentially uniform directional and frequency dependence over a speech frequency range, is scalable and robust against small incorrect positioning of the capsules, fifteen or twenty-one microphone capsules (K15,11-K15,35, K21,11-K21,37) are arranged on a carrier such that they lie on three similar branches, each with the same number of microphone capsules, which are rotated against each other by 120°. Each of the microphone capsules lies on a corner of a triangle of a grid in a flat isometric coordinate system with three axes rotated by 120° against each other and forming the grid of equilateral triangles.

Abstract: Microphone arrays comprise several microphone capsules, the outputs of which being electronically combined for directional recording of sound. The directional and frequency properties of the microphone array depend on the number and positions of the microphone array. In order to obtain the smallest possible microphone array with only few microphone capsules, which, however, has an essentially uniform directional and frequency dependence over a speech frequency range, is scalable and robust against small incorrect positioning of the capsules, fifteen or twenty-one microphone capsules (K15,11-K15,35, K21,11-K21,37) are arranged on a carrier such that they lie on three similar branches, each with the same number of microphone capsules, which are rotated against each other by 120°. Each of the microphone capsules lies on a corner of a triangle of a grid in a flat isometric coordinate system with three axes rotated by 120° against each other and forming the grid of equilateral triangles.

Abstract: Aspects of the present disclosure are directed to enabling viewing and controlling a mixed reality capture feed from within an artificial reality environment. Additional aspects of the present disclosure are directed to applying a rotated viewing angle to a wearable device UI, interpreting input according to the rotated viewing angle, and providing corner controls. Further aspects of the present disclosure are directed to customizing elements of an artificial reality environment based on recognition of a user's object-of-focus.

Abstract: During telephone calls in a public space, users may be reluctant to provide private or secret information, due to the risk of eavesdropping. A hands-free solution for entering secret information to electronic speech communication devices is based on speech processing. A method for speech processing a voice input data stream comprises steps of scanning the voice input data stream and detecting a spoken delimiter therein, determining a predefined audio sample corresponding to the detected spoken delimiter, inserting the determined predefined audio sample into the voice input data stream at the spoken delimiter, wherein a substituted voice data stream is obtained and wherein speech portions of the voice input data stream at least before the spoken delimiter remain in the substituted voice data stream, and providing the substituted voice data stream for output towards a recipient.

Abstract: During telephone calls in a public space, users may be reluctant to provide private or secret information, due to the risk of eavesdropping. A hands-free solution for entering secret information to electronic speech communication devices is based on speech processing. A method for speech processing a voice input data stream comprises steps of scanning the voice input data stream and detecting a spoken delimiter therein, determining a predefined audio sample corresponding to the detected spoken delimiter, inserting the determined predefined audio sample into the voice input data stream at the spoken delimiter, wherein a substituted voice data stream is obtained and wherein speech portions of the voice input data stream at least before the spoken delimiter remain in the substituted voice data stream, and providing the substituted voice data stream for output towards a recipient.

Abstract: Microphone arrays with automatic beam focusing may easily focus on disturbing sound sources. In order to prevent this unwanted behavior, a predefined control sound signal is replayed from a direction of a disturbing sound source. The microphone array detects the predefined control sound signal, determines the direction of replay and in response performs a re-configuration according to the control sound signal. The reconfiguration may comprise eliminating the direction from its scanning range or cancel a previously made elimination of a different direction.

Abstract: Disclosed are methods and corresponding systems for audio processing of audio signals after applying a noise reduction procedure such as noise cancellation and/or noise suppression, according to various embodiments. A method may include calculating spectral envelopes for corresponding samples of an initial audio signal and the audio signal transformed by application of the noise cancellation and/or suppression procedure. Multiple spectral envelope interpolations may be calculated between these two spectral envelopes. The interpolations may be compared to predetermined reference spectral envelopes associated with predefined clean reference speech. One of the generated interpolations, which is the closest to one of the predetermined reference spectral envelopes, may be selected. The selected interpolation may be used for restoration of the transformed audio signal such that at least a part of the frequency spectrum of the transformed audio signal is modified to the levels of the selected interpolation.

Abstract: Audio signal bandwidth extension may be performed on a narrow bandwidth signal received from a remote source over the audio communication network. The narrow band signal bandwidth may be extended such that the bandwidth is greater than that of the audio communication network. The signal may be extended by synthesizing an audio signal having spectral values within an extended bandwidth from synthetic components. The synthetic components may be generated using parameters derived from original narrowband audio signal. The audio signal may be synthesized in the form of an excitation signal and vocal tract envelope. The excitation signal and vocal tract may be extended independently. In various embodiments, excitation components may be derived from constrained synthesis using a constraint filter with nulls in regions where the extension is desired.

Abstract: In accordance with the present invention, computer implemented methods and systems are provided for representing and modeling the temporal structure of audio signals. In response to receiving a signal, a time-to-frequency domain transformation on at least a portion of the received signal to generate a frequency domain representation is performed. The time-to-frequency domain transformation converts the signal from a time domain representation to the frequency domain representation. A frequency domain linear prediction (FDLP) is performed on the frequency domain representation to estimate a temporal envelope of the frequency domain representation. Based on the temporal envelope, one or more speech features are generated.

Abstract: In accordance with the present invention, computer implemented methods and systems are provided for representing and modeling the temporal structure of audio signals. In response to receiving a signal, a time-to-frequency domain transformation on at least a portion of the received signal to generate a frequency domain representation is performed. The time-to-frequency domain transformation converts the signal from a time domain representation to the frequency domain representation. A frequency domain linear prediction (FDLP) is performed on the frequency domain representation to estimate a temporal envelope of the frequency domain representation. Based on the temporal envelope, one or more speech features are generated.

Abstract: In accordance with the present invention, computer implemented methods and systems are provided for representing and modeling the temporal structure of audio signals. In response to receiving a signal, a time-to-frequency domain transformation on at least a portion of the received signal to generate a frequency domain representation is performed. The time-to-frequency domain transformation converts the signal from a time domain representation to the frequency domain representation. A frequency domain linear prediction (FDLP) is performed on the frequency domain representation to estimate a temporal envelope of the frequency domain representation. Based on the temporal envelope, one or more speech features are generated.

Abstract: Systems and methods are disclosed for searching or finding music with music, by searching, e.g., for music from a library that has a sound that is similar to a given sound provided as a search query, and to methods and systems for tracking revenue generated by these computer-user interactions, and for promoting music and selling advertising space. These include, inter alia, systems that allow a user to discover unknown music, and systems that allow a user to look for music based directly on queries formed from sounds that the user likes. In some embodiments these queries are comprised of a clip or relatively small segment of a larger media file. A client server system comprising web graphical elements, advertisements and/or other affiliated revenue links, elements in support of the music query and a music player, a database, elements for matching music clips to clips from a library, and elements to present results.

Abstract: This invention enables content-based GUI modification. The invention allows a new way of packaging elements of the GUI together with the application content to enable only the GUI necessary to present the received content. The packaged elements are then transmitted together with such content.