Abstract: A method, a device and a computer-readable storage medium with instructions for processing sensor data. In a first step, camera images are taken by a camera. Additionally, 3D-checkpoints are detected by at least one 3D-sensor. Optionally, at least one of the camera images can be segmented. The camera images are then merged with the 3D-checkpoints by a data fusion circuit to form data of a virtual sensor. The resulting data are finally output for further processing.

Abstract: An orthodontic apparatus (1; 10) is designed with at least one orthodontically effective element (4; 12) and at least one holding element (5; 13) for holding the apparatus (1; 10) on the jaw, whereby the apparatus (1;10) is designed in one piece. In a method for producing an orthodontic apparatus, an electronic data set is produced for a three-dimensional model (2) of a jaw to which the apparatus is supposed to be attached. Further, a virtual positioning of one or more orthodontic elements (4; 12) of the apparatus on jaw model (2) takes place. An electronic data set of a three-dimensional model (3; 11) of the apparatus is subsequently produced. The apparatus (1; 10) is then produced by an additive and/or removal production method according to the data set of the apparatus model.

Abstract: An orthodontic apparatus (1; 10) is designed with at least one orthodontically effective element (4; 12) and at least one holding element (5; 13) for holding the apparatus (1; 10) on the jaw, whereby the apparatus (1; 10) is designed in one piece. In a method for producing an orthodontic apparatus, an electronic data set is produced for a three-dimensional model (2) of a jaw to which the apparatus is supposed to be attached. Further, a virtual positioning of one or more orthodontic elements (4; 12) of the apparatus on jaw model (2) takes place. An electronic data set of a three-dimensional model (3; 11) of the apparatus is subsequently produced. The apparatus (1; 10) is then produced by an additive and/or removal production method according to the data set of the apparatus model.

Abstract: A method, computer program product, and computer system for identifying, by a computing device, a plurality of sources. One or more feature values of a plurality of features may be assigned to a first source of the plurality of sources. One or more feature values of the plurality of features may be assigned to a second source of the plurality of sources. A first score for the first source and a second score for the second source may be determined based upon, at least in part, the one or more feature values assigned to the first source and the second source. One of the first source and the second source may be selected for spatial processing based upon, at least in part, the first score for the first source and the second score for the second source.

Abstract: A low-complexity method and apparatus for detection of voiced speech and pitch estimation is disclosed that is capable of dealing with special constraints given by applications where low latency is required, such as in-car communication (ICC) systems. An example embodiment employs very short frames that may capture only a single excitation impulse of voiced speech in an audio signal. A distance between multiple such impulses, corresponding to a pitch period, may be determined by evaluating phase differences between low-resolution spectra of the very short frames. An example embodiment may perform pitch estimation directly in a frequency domain based on the phase differences and reduce computational complexity by obviating transformation to a time domain to perform the pitch estimation. In an event the phase differences are determined to be substantially linear, an example embodiment enhances voice quality of the voiced speech by applying speech enhancement to the audio signal.

Abstract: A low-complexity method and apparatus for detection of voiced speech and pitch estimation is disclosed that is capable of dealing with special constraints given by applications where low latency is required, such as in-car communication (ICC) systems. An example embodiment employs very short frames that may capture only a single excitation impulse of voiced speech in an audio signal. A distance between multiple such impulses, corresponding to a pitch period, may be determined by evaluating phase differences between low-resolution spectra of the very short frames. An example embodiment may perform pitch estimation directly in a frequency domain based on the phase differences and reduce computational complexity by obviating transformation to a time domain to perform the pitch estimation. In an event the phase differences are determined to be substantially linear, an example embodiment enhances voice quality of the voiced speech by applying speech enhancement to the audio signal.

Abstract: Systems and methods are introduced to perform noise suppression of an audio signal. The audio signal includes foreground speech components and background noise. The foreground speech components correspond to speech from a user's speaking into an audio receiving device. The background noise includes babble noise that includes speech from one or more interfering speakers. A soft speech detector determines, dynamically, a speech detection result indicating a likelihood of a presence of the foreground speech components in the audio signal. The speech detection result is employed to control, dynamically, an amount of attenuation of the noise suppression to reduce the babble noise in the audio signal. Further processing achieves a more stationary background and reduction of musical tones in the audio signal.

Abstract: A method, computer program product, and computer system for identifying, by a computing device, a plurality of sources. One or more feature values of a plurality of features may be assigned to a first source of the plurality of sources. One or more feature values of the plurality of features may be assigned to a second source of the plurality of sources. A first score for the first source and a second score for the second source may be determined based upon, at least in part, the one or more feature values assigned to the first source and the second source. One of the first source and the second source may be selected for spatial processing based upon, at least in part, the first score for the first source and the second score for the second source.

Abstract: A method, computer program product, and computer system for identifying, by a computing device, a plurality of sources, wherein a first source of the plurality of sources is a source of interest and wherein a second source of the plurality of sources is an interference source. The first source and the second source may be monitored simultaneously by implementing a spatial pre-filter for acoustic source localization.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a first signal emitted from one or more sources. A second signal may be received emitted from the one or more sources. A first confidence level that the wake-up-word is included in the first signal may be determined. A second confidence level that the wake-up-word is included in the second signal may be determined. It may be identified that the wake-up-word originated from a first source of the one or more sources based upon, at least in part, the first and second confidence levels. The first source may be enabled to participate in a dialog phase.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a speech signal from a speaker via a plurality of microphone zones. A temporal cue based confidence may be determined for at least a portion of the plurality of microphone zones. A power cue based confidence may be determined for at least a portion of the plurality of microphone zones. A microphone zone of the plurality of microphone zones from which to use an output signal of the speaker may be identified based upon, at least in part, a combination of the temporal cue based confidence and the power cue based confidence.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a first signal emitted from one or more sources. A second signal may be received emitted from the one or more sources. A first confidence level that the wake-up-word is included in the first signal may be determined. A second confidence level that the wake-up-word is included in the second signal may be determined. It may be identified that the wake-up-word originated from a first source of the one or more sources based upon, at least in part, the first and second confidence levels. The first source may be enabled to participate in a dialogue phase.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a speech signal from a speaker via a plurality of microphone zones. A temporal cue based confidence may be determined for at least a portion of the plurality of microphone zones. A power cue based confidence may be determined for at least a portion of the plurality of microphone zones. A microphone zone of the plurality of microphone zones from which to use an output signal of the speaker may be identified based upon, at least in part, a combination of the temporal cue based confidence and the power cue based confidence.

Abstract: Speech processing methods may rely on voice activity detection (VAD) that separates speech from noise. Example embodiments of a computationally low complex VAD feature that is robust against various types of noise is introduced. By considering an alternating excitation structure of low and high frequencies, speech is detected with a high confidence. The computationally low complex VAD feature can cope even with the limited spectral resolution that may be typical for a communication system, such as an in-car-communication (ICC) system. Simulation results confirm the robustness of the computationally low complex VAD feature and show an increase in performance relative to established VAD features.

Abstract: Methods and apparatus to process microphone signals by a speech enhancement module to generate an audio stream signal including first and second metadata for use by a speech recognition module. In an embodiment, speech recognition is performed using endpointing information including transitioning from a silence state to a maybe speech state, in which data is buffered, based on the first metadata and transitioning to a speech state, in which speech recognition is performed, based upon the second metadata.

Abstract: Systems and methods are introduced to perform noise suppression of an audio signal. The audio signal includes foreground speech components and background noise. The foreground speech components correspond to speech from a user's speaking into an audio receiving device. The background noise includes babble noise that includes speech from one or more interfering speakers. A soft speech detector determines, dynamically, a speech detection result indicating a likelihood of a presence of the foreground speech components in the audio signal. The speech detection result is employed to control, dynamically, an amount of attenuation of the noise suppression to reduce the babble noise in the audio signal. Further processing achieves a more stationary background and reduction of musical tones in the audio signal.

Abstract: A method, computer program product, and computer system for receiving, by a computing device, a first signal emitted from one or more sources. A second signal may be received emitted from the one or more sources. A first confidence level that the wake-up-word is included in the first signal may be determined. A second confidence level that the wake-up-word is included in the second signal may be determined. It may be identified that the wake-up-word originated from a first source of the one or more sources based upon, at least in part, the first and second confidence levels. The first source may be enabled to participate in a dialog phase. The second source may be excluded from participating in the dialog phase.

Abstract: Methods and apparatus to process microphone signals by a speech enhancement module to generate an audio stream signal including first and second metadata for use by a speech recognition module. In an embodiment, speech recognition is performed using endpointing information including transitioning from a silence state to a maybe speech state, in which data is buffered, based on the first metadata and transitioning to a speech state, in which speech recognition is performed, based upon the second metadata.

Abstract: An orthodontic apparatus (1; 10) is designed with at least one orthodontically effective element (4; 12) and at least one holding element (5; 13) for holding the apparatus (1; 10) on the jaw, whereby the apparatus (1;10) is designed in one piece. In a method for producing an orthodontic apparatus, an electronic data set is produced for a three-dimensional model (2) of a jaw to which the apparatus is supposed to be attached. Further, a virtual positioning of one or more orthodontic elements (4; 12) of the apparatus on jaw model (2) takes place. An electronic data set of a three-dimensional model (3; 11) of the apparatus is subsequently produced. The apparatus (1; 10) is then produced by an additive and/or removal production method according to the data set of the apparatus model.

Abstract: An elastic coupling, in particular an elastic shaft coupling, includes a metallic outer part, a metallic inner part and elastic buffers arranged so as to be spaced apart between the outer part and the inner part. The outer part includes radially inwardly projecting elevations which define buffer contact surfaces and the inner part includes radially outwardly projecting elevations which define buffer contact surfaces. In order for the elastic coupling to allow for gentler startup and an operating behaviour optimized in terms of rotary oscillation in comparison with conventional couplings of this type. The elastic buffers are arranged in at least two groups which are radially spaced apart. At least one annular intermediate part is arranged between the adjacent buffer groups and is connected torsionally to both the outer part and to the inner part via the elastic buffers.