Abstract: Presented herein are techniques for generated an integrated estimate of a target sound (e.g., speech) in sound signals received by at least a local microphone array of a device. In embodiments, the integrated estimate may be generated based on sound signals received by the at least a local microphone array of a device and at least one external microphone.

Abstract: Presented herein are techniques for generated an integrated estimate of a target sound (e.g., speech) in sound signals received by at least a local microphone array of a device. In embodiments, the integrated estimate may be generated based on sound signals received by the at least a local microphone array of a device and at least one external microphone.

Abstract: A method of active noise equalization in a system comprising a plurality of nodes is disclosed, wherein each node comprises at least one acoustic sensor and at least one acoustic actuator, and each node has an associated target spectral noise profile and an associated set of adaptive filter coefficients.

Abstract: An audio-processing device having an audio input, for receiving audio signals, each audio signal having a mixture of components, each corresponding to a sound source, and a control input, for receiving, for each sound source, a desired gain factor associated with the source, by which it is desired to amplify the corresponding component, and an auxiliary signal generator, for generating at least one auxiliary signal from the audio signals, and with a different mixture of components as compared with a reference audio signal; and a scaling coefficient calculator, for calculating scaling coefficients based upon the desired gain factors and upon parameters of the different mixture, each scaling coefficient associated with one of the auxiliary signal and optionally the reference audio signal, and an audio synthesis unit, for synthesizing an output audio signal by applying scaling coefficients to the auxiliary signal and optionally the reference audio signal and combining the results.

Abstract: Various embodiments for components and associated methods that can be used in a binaural speech enhancement system are described. The components can be used, for example, as a pre-processor for a hearing instrument and provide binaural output signals based on binaural sets of spatially distinct input signals that include one or more input signals. The binaural signal processing can be performed by at least one of a binaural spatial noise reduction unit and a perceptual binaural speech enhancement unit. The binaural spatial noise reduction unit performs noise reduction while preferably preserving the binaural cues of the sound sources. The perceptual binaural speech enhancement unit is based on auditory scene analysis and uses acoustic cues to segregate speech components from noise components in the input signals and to enhance the speech components in the binaural output signals.

Abstract: In one aspect of the present invention, a method to reduce noise in a noisy speech signal is disclosed The method comprises: applying at least two versions of the noisy speech signal to a first filter, whereby that first filter outputs a speech reference signal and at least one noise reference signal, applying a filtering operation to each of the at least one noise reference signals, and subtracting from the speech reference signal each of the filtered noise reference signals, wherein the filtering operation is performed with filters having filter coefficients determined by taking into account speech leakage contributions in the at least one noise reference signal.

Abstract: The binaural hearing system (1) comprises ITF means (3a;3b) for providing at least one interaural transfer function (30a;30b); noise reduction means (5a;5b) for performing noise reduction in dependence of said at least one interaural transfer function. The method of operating a binaural hearing system (1) comprises the steps of providing at least one interaural transfer function (30a;30b); performing noise reduction in dependence of said at least one interaural transfer function. Preferably, said noise reduction means (5a;5b) comprises two binaural Wiener filters (5a,5b) each having a cost function comprising at least one term describing a desired interaural transfer function, wherein said at least one interaural transfer function provided by said ITF means (3a,3b) is assigned to said at least one term. Preferably, said cost function comprises a speech distortion term, a residual noise term and two ITF terms for preserving the interaural transfer functions of speech and noise components.

Abstract: Various embodiments for components and associated methods that can be used in a binaural speech enhancement system are described. The components can be used, for example, as a pre-processor for a hearing instrument and provide binaural output signals based on binaural sets of spatially distinct input signals that include one or more input signals. The binaural signal processing can be performed by at least one of a binaural spatial noise reduction unit and a perceptual binaural speech enhancement unit. The binaural spatial noise reduction unit performs noise reduction while preferably preserving the binaural cues of the sound sources. The perceptual binaural speech enhancement unit is based on auditory scene analysis and uses acoustic cues to segregate speech components from noise components in the input signals and to enhance the speech components in the binaural output signals.

Abstract: A method and system for retrieving a desired user data symbol sequence from a received signal are disclosed. In one embodiment, the method includes i) receiving a channel modified version of a transmitted signal comprising a plurality of user data symbol sequences, each being encoded with a user specific known code, ii) determining an equalization filter directly and in a deterministic way from the received signal and iii) applying the equalization filter on the received signal to thereby retrieve the transmitted signal.

Abstract: A method of determining IQ imbalance introduced on an RF multicarrier signal received via a channel on a direct conversion analog receiver is disclosed. In one embodiment, the method comprises i) receiving a training signal on the receiver, ii) demodulating the training signal on the receiver, iii) estimating a first frequency domain channel characteristic of the channel based on the demodulated training signal, iv) defining a predetermined relationship between a corrected frequency domain channel characteristic of the channel and the first channel characteristic, the predetermined relationship comprising at least one IQ imbalance parameter, and v) determining the at least one IQ imbalance parameter such that the corrected channel characteristic satisfies a channel constraint.

Abstract: A method and system for retrieving a desired user data symbol sequence from a received signal are disclosed. In one embodiment, the method includes i) receiving a channel modified version of a transmitted signal comprising a plurality of user data symbol sequences, each being encoded with a user specific known code, ii) determining an equalization filter directly and in a deterministic way from the received signal and iii) applying the equalization filter on the received signal to thereby retrieve the transmitted signal.

Abstract: In one aspect of the present invention, a method to reduce noise in a noisy speech signal is disclosed The method comprises applying at least two versions of the noisy speech signal to a first filter, whereby that first filter outputs a speech reference signal and at least one noise reference signal, applying a filtering operation to each of the at least one noise reference signals, and subtracting from the speech reference signal each of the filtered noise reference signals, wherein the filtering operation is performed with filters having filter coefficients determined by taking into account speech leakage contributions in the at least one noise reference signal.

Abstract: The present invention is related to a system and method for wideband multiple access telecommunication. In the method, a block is transmitted from a basestation to a terminal. The block comprises a plurality of chip symbols scrambled with a base station specific scrambling code, the plurality of chip symbols comprising a plurality of spread user specific data symbols which are user specific data symbols spread by using user specific spreading codes and at least one pilot symbol. In the terminal, at least two independent signals that comprise at least a channel distorted version of the transmitted block are generated. The two independent signals are combined with a combiner filter with filter coefficients which are determined by using the pilot symbol, thus a combined filtered signal is obtained. The combined filtered signal is despread and descrambled with a composite code of the basestation specific scrambling code and one of the user specific codes.

Abstract: A method for determining at least one operational transmit power Snk over a communication channel (Ln) coupled to a disturbance causing transceiver (n) for at least one tone (k), comprises the steps of provision of at least one reference victim communication channel (Lref) for representing at least one reference victim for all victim communication channels which are degraded by cross-talk interferences by said communication channel (Ln), and maximizing the data rate (Rref) over said at least one reference victim communication channel (Lref) under the constraint that the data rate over said communication channel (Ln) achieves a target rate (Rntarget) without exceeding a power constraint (Pnmax).

Abstract: A method of determining IQ imbalance introduced on an RF multicarrier signal received via a channel on a direct conversion analog receiver is disclosed. In one embodiment, the method comprises i) receiving a training signal on the receiver, ii) demodulating the training signal on the receiver, iii) estimating a first frequency domain channel characteristic of the channel based on the demodulated training signal, iv) defining a predetermined relationship between a corrected frequency domain channel characteristic of the channel and the first channel characteristic, the predetermined relationship comprising at least one IQ imbalance parameter, and v) determining the at least one IQ imbalance parameter such that the corrected channel characteristic satisfies a channel constraint.

Abstract: The present invention is related to a system and method for wideband multiple access telecommunication. In the method, a block is transmitted from a basestation to a terminal. The block comprises a plurality of chip symbols scrambled with a base station specific scrambling code, the plurality of chip symbols comprising a plurality of spread user specific data symbols which are user specific data symbols spread by using user specific spreading codes and at least one pilot symbol. In the terminal, at least two independent signals that comprise at least a channel distorted version of the transmitted block are generated. The two independent signals are combined with a combiner filter with filter coefficients which are determined by using the pilot symbol, thus a combined filtered signal is obtained. The combined filtered signal is despread and descrambled with a composite code of the basestation specific scrambling code and one of the user specific codes.