Abstract: A network architecture for a three-dimensional object detection in a point cloud to a method and to vehicle. The network architecture for a three-dimensional object detection in a point cloud may include an encoder, a backbone, and a head, wherein the backbone and/or the head may include a three-dimensional convolution component for processing three-dimensional data of the point cloud within a three-dimensional voxel grid.

Abstract: A method, system and computer program for processing image data, to a vehicle comprising such a system, and to a method, system and computer program for generating a filter. Image data processing may include obtaining the image data, and applying a filter on the image data to generate filtered image data, the filter being configured to suppress adversarial perturbations within the image data. The filtered image data is processed using a machine-learning model.

Abstract: A signal processor comprising: a signal-manipulation-block configured to: receive a cepstrum-input-signal, wherein the cepstrum-input-signal is in the cepstrum domain and comprises a plurality of bins; receive a pitch-bin-identifier that is indicative of a pitch-bin in the cepstrum-input-signal; and generate a cepstrum-output-signal based on the cepstrum-input-signal by: scaling the pitch-bin relative to one or more of the other bins of the cepstrum-input-signal; or determining an output-pitch-bin-value based on the pitch-bin, and setting one or more of the other bins of the cepstrum-input-signal to a predefined value; or determining an output-other-bin-value based on one or more of the other bins of the cepstrum-input-signal, and setting the pitch-bin to a predefined value.

Abstract: A speech-signal-processing-circuit configured to receive a time-frequency-domain-reference-speech-signal and a time-frequency-domain-degraded-speech-signal. The time-frequency-domain-reference-speech-signal comprises: an upper-band-reference-component with frequencies that are greater than a frequency-threshold-value; and a lower-band-reference-component with frequencies that are less than the frequency-threshold-value. The time-frequency-domain-degraded-speech-signal comprises: an upper-band-degraded-component with frequencies that are greater than the frequency-threshold-value; and a lower-band-degraded-component with frequencies that are less than the frequency-threshold-value.

Abstract: A signal processor comprising: a signal-manipulation-block configured to: receive a cepstrum-input-signal, wherein the cepstrum-input-signal is in the cepstrum domain and comprises a plurality of bins; receive a pitch-bin-identifier that is indicative of a pitch-bin in the cepstrum-input-signal; and generate a cepstrum-output-signal based on the cepstrum-input-signal by: scaling the pitch-bin relative to one or more of the other bins of the cepstrum-input-signal; or determining an output-pitch-bin-value based on the pitch-bin, and setting one or more of the other bins of the cepstrum-input-signal to a predefined value; or determining an output-other-bin-value based on one or more of the other bins of the cepstrum-input-signal, and setting the pitch-bin to a predefined value.

Abstract: A speech-signal-processing-circuit configured to receive a time-frequency-domain-reference-speech-signal and a time-frequency-domain-degraded-speech-signal. The time-frequency-domain-reference-speech-signal comprises: an upper-band-reference-component with frequencies that are greater than a frequency-threshold-value; and a lower-band-reference-component with frequencies that are less than the frequency-threshold-value. The time-frequency-domain-degraded-speech-signal comprises: an upper-band-degraded-component with frequencies that are greater than the frequency-threshold-value; and a lower-band-degraded-component with frequencies that are less than the frequency-threshold-value.

Abstract: A hearing system for improving intelligibility during the holding of a telephone conversation, includes a receiving device for receiving a first signal having first acoustic information and converting the first signal into electrical signals having the first acoustic information, a signal processing device for processing the electrical signals having the first acoustic information into electrical signals having second acoustic information, and an output device for outputting a second signal having the second acoustic information. A method for improving intelligibility during the holding of a telephone conversation using a hearing system is also provided.

Abstract: A hearing system for improving intelligibility during the holding of a telephone conversation, includes a receiving device for receiving a first signal having first acoustic information and converting the first signal into electrical signals having the first acoustic information, a signal processing device for processing the electrical signals having the first acoustic information into electrical signals having second acoustic information, and an output device for outputting a second signal having the second acoustic information. A method for improving intelligibility during the holding of a telephone conversation using a hearing system is also provided.

Abstract: Speaker-dependent speech recognition is performed upon detecting a speech signal encompassing a voice command. The speech signal is divided into time frames and characterized in each detected time frame by forming a corresponding property vector. A language-independent feature vector sequence is formed from one or several property vectors and then stored. The language-independent feature vector sequence is allocated to a language-dependent sequence of model vectors in a speech resource having a plurality of model vectors. A piece of allocation information indicating allocation of the language-independent feature vector sequence to a language-dependent sequence of model vectors is stored, then the voice command allocated to the model vector sequence is identified.

Abstract: A speech signal is processed for subsequent speech recognition. The speech signal is tainted by noise and represents at least one speech command. The following steps are executed: a) recording of the noise-tainted speech signal; b) use of noise reduction on the speech signal to generate a noise-reduced speech signal; c) normalization of the noise-reduced speech signal to a target signal value with the aid of a normalization factor, to generate a noise-reduced, normalized speech signal).

Abstract: A method reduces for noise in a speech input signal of a speaker by detecting the speech input signal; accessing a determined speech characteristic of the speaker; reducing a noise portion in the speech input signal using the determined speech characteristic of the speaker.

Abstract: Speaker-dependent speech recognition is performed upon detecting a speech signal encompassing a voice command. The speech signal is divided into time frames and characterized in each detected time frame by forming a corresponding property vector. A property vector sequence is formed from one or several property vectors and then stored. The property vector sequence is allocated to a sequence of model vectors in a speech resource having a plurality of model vectors. A piece of allocation information indicating allocation of the property vector sequence to a sequence of model vectors is stored, then the voice command allocated to the model vector sequence is identified.

Abstract: Disclosed is a method for suppressing (10) echo (z(t)) in uplink data (y(t), 12-16) originating from a terminal (2, 3), said method allowing a delay and a decrease in quality during echo suppression of uplink data originating from a wireless terminal (2, 3). According to the inventive method, original or copied downlink data (12-16) and uplink data (19-21) are analyzed in order to prepare echo suppression (10), and uplink data (19-21) is modified in the compressed state in order to reduce echo (10) by using the results of said analysis (9) of the downlink data (12-16) and uplink data (19-21).

Abstract: The invention relates to a method for encoding voice signals, especially so-called voice onset sections. By establishing the first amplification factor, the data quantity for representing the whole of the first or adaptive amplification factor and adaptive code book entry is reduced, whereby other parameters which occur during the voice encoding can be represented in a more precise manner. The invention also relates to a method for transmitting voice signals which are encoded in such a way.

Abstract: A method and device are provided for automatically differentiating and/or detecting acoustic signals, whereby the signals are statistically analyzed, at least in part, and their reflection coefficients of at least one are calculated. Thereafter, a comparison value, which is dependent exclusively on a single reflection coefficient, is calculated and compared with at least one predetermined reference value.

Abstract: A method is provided for estimating a codec parameter, wherein the method is particularly applicable to the estimation of filter coefficients in all known forms thereof, from amplification factors and from speech base frequency values such as occur in the coding of speech. Extrapolation, interpolation and linear prediction are used in combination for the estimation.

Abstract: With a method and a device for error masking in digital transmission systems whereby a distribution function for estimating transmitted parameters is determined at the receiving end, the method provides that a distribution function is adjusted around the output signal value of a predictor and is integrated by sections into a new distribution function; this new distribution function is multiplied by a distribution function which takes the received quality into consideration and the result of an a posteriori distribution which can be used with conventional estimation methods for the final parameter estimation.