Abstract: An ultrasound image compounding method that is suitable for use with an ICE catheter that includes a first ultrasound transceiver array and a second ultrasound transceiver array, the first ultrasound transceiver array and the second ultrasound transceiver array being axially separated along a length of the ICE catheter, is described. In the method, first array data corresponding to ultrasound signals detected by the first ultrasound transducer array in response to an insonification of a region of interest by the first ultrasound transducer array at a first insonification angle; and second array data corresponding to ultrasound signals detected by the second ultra- sound transducer array in response to an insonification of the region of interest by the second ultrasound transducer array at a second insonification angle; are received. A compound image corresponding to the region of interest is generating based on the first array data and the second array data.

Abstract: An audio capture apparatus comprises a first beamformer (303) which is arranged to generate a beamformed audio output signal. An adapter (305) adapts beamform parameters of the first beamformer and a detector (307) detects an attack of speech in the beamformed audio output signal. A controller (309) controls the adaptation of the beamform parameters to occur in a predetermined adaptation time interval determined in response to the detection of the attack of speech. The beamformer (303) may generate noise reference signal(s) and the detector (309) may be arranged to detect the attack of speech in response to a comparison of a signal level of the beamformed audio output signal relative to a signal level of the at least one noise reference signal.

Abstract: An audio capture apparatus comprises a first beamformer (303) which is arranged to generate a beamformed audio output signal. An adapter (305) adapts beamform parameters of the first beamformer and a detector (307) detects an attack of speech in the beamformed audio output signal. A controller (309) controls the adaptation of the beamform parameters to occur in a predetermined adaptation time interval determined in response to the detection of the attack of speech. The beamformer (303) may generate noise reference signal(s) and the detector (309) may be arranged to detect the attack of speech in response to a comparison of a signal level of the beamformed audio output signal relative to a signal level of the at least one noise reference signal.

Abstract: An audio capture apparatus comprises a microphone array (301) and a beamformer (303) arranged to generate a beamformed audio output signal and a noise reference signal. A first and second transformer (309, 311) generates a first and second frequency domain signal from a frequency transform of the beamformed audio output signal and noise reference signal respectively. A difference processor (313) generates time frequency tile difference measures which for a given frequency is indicative of a difference between a monotonic function of a norm (magnitude) of a time frequency tile value of the first frequency domain signal and a monotonic function of a norm of a time frequency tile value of the second frequency domain signal for the first frequency. An estimator (315) generates an estimate indicative of whether the audio output signal comprises a point audio source in response to a combined difference value for time frequency tile difference measures for frequencies above a frequency threshold.

Abstract: An apparatus for capturing audio comprises a first beamformer (305) coupled to a microphone array (301) and arranged to generate a first beamformed audio output. A plurality of constrained beamformers (309, 311) each generates a constrained beamformed audio output. A first adapter (307) adapts beamform parameters of the first beamformer (305) and a second adapter (313) adapts constrained beamform parameters for the plurality of constrained beamformers (309, 311). A difference processor (317) determines a difference measure for the constrained beamformers (309, 311) where the difference measure is indicative of the difference between beams formed by the first beamformer (305) and the constrained beamformers (309, 311).

Abstract: An apparatus for capturing audio comprises a first beamformer (305) coupled to a microphone array (301) and arranged to generate a first beamformed audio output. A plurality of constrained beamformers (309, 311) each generates a constrained beamformed audio output. A first adapter (307) adapts beamform parameters of the first beamformer (305) and a second adapter (313) adapts constrained beamform parameters for the plurality of constrained beamformers (309, 311). A difference processor (317) determines a difference measure for the constrained beamformers (309, 311) where the difference measure is indicative of the difference between beams formed by the first beamformer (305) and the constrained beamformers (309, 311).

Abstract: A beamforming audio capture apparatus comprises a microphone array (301) which is coupled to a first beamformer (303) and a second beamformer (305). The beamformers (303, 305) are filter-and-combine beamformers comprising a plurality of beamform filters each having an adaptive impulse response. A difference processor (309) determines a difference measure between beams of the first beamformer (303) and the second beamformer (305) in response to a comparison of the adaptive impulse responses of the two beamformers (303, 305). The difference measure may e.g. be used to combine the output signals of the beamformers (303, 305). An improved difference measure less sensitive to e.g. diffuse noise may be provided.

Abstract: A beamforming audio capture apparatus comprises a microphone array (301) which is coupled to a first beamformer (303) and a second beamformer (305). The beamformers (303, 305) are filter-and-combine beamformers comprising a plurality of beamform filters each having an adaptive impulse response. A difference processor (309) determines a difference measure between beams of the first beamformer (303) and the second beamformer (305) in response to a comparison of the adaptive impulse responses of the two beamformers (303, 305). The difference measure may e.g. be used to combine the output signals of the beamformers (303, 305). An improved difference measure less sensitive to e.g. diffuse noise may be provided.

Abstract: An audio signal processing apparatus comprises a receiver (403) receiving an audio signal sampled at a first sampling frequency, the audio signal having a maximum frequency below half the first sampling frequency by a first frequency margin. A filter bank (405) generates subband signals for the digital audio signal using overlapping sub-filters. A first frequency shifter (407) applies a frequency shift to at least one subband of the set of subbands and a decimator (409) decimates the subband signals by a decimation factor resulting in a decimated sampling frequency being at least twice a bandwidth of each of the overlapping sub-filters. The frequency shift for a subband is arranged to shift the subband to a frequency interval being a multiple of a frequency interval from zero to half the decimated sample frequency. The subband may be individually processed and the processed subbands may subsequently be combined to generate a full band output signal.

Abstract: An audio capture apparatus comprises a microphone array (301) and a beamformer (303) arranged to generate a beamformed audio output signal and a noise reference signal. A first and second transformer (309, 311) generates a first and second frequency domain signal from a frequency transform of the beamformed audio output signal and noise reference signal respectively. A difference processor (313) generates time frequency tile difference measures which for a given frequency is indicative of a difference between a monotonic function of a norm (magnitude) of a time frequency tile value of the first frequency domain signal and a monotonic function of a norm of a time frequency tile value of the second frequency domain signal for the first frequency. An estimator (315) generates an estimate indicative of whether the audio output signal comprises a point audio source in response to a combined difference value for time frequency tile difference measures for frequencies above a frequency threshold.

Abstract: An inhaler with a housing (H) comprising an air-inlet (A_I) and a an air-outlet (A_O). Inside the housing (H) a flow path (FP) is defined between air-inlet (A_I) and air-outlet (A_O) where a dispenser (DP) is arranged to dispense an aerosol or a dry powder in the flow path (FP). Two sensors (S1, S2), e.g. microphones, are positioned spaced apart at external surfaces of the housing (H) to sense sound or vibrations resulting from a flow in the flow path (FP) at two different positions. This allows a precise detection of flow velocity during inhalation based on the sound or vibrations sensed by the two sensors (S1, S2), thus allowing examination of correct use of the inhaler. Further, the use of two spaced apart sensors (S1, S2) facilitates identification of priming and firing events in the sensed sound or vibrations which may also be used in evaluating the use of the inhaler.

Abstract: An audio signal processing apparatus comprises a receiver (403) receiving an audio signal sampled at a first sampling frequency, the audio signal having a maximum frequency below half the first sampling frequency by a first frequency margin. A tilter bank (405) generates subband signals for the digital audio signal using overlapping sub-filters. A first frequency shifter (407) applies a frequency shift to at least one subband of the set of subbands and a decimator (409) decimates the subband signals by a N decimation factor resulting in a decimated sampling frequency being at least twice a bandwidth of each of the overlapping sub-filters. The frequency shift for a subband is arranged to shift the subband to a frequency interval being a multiple of a frequency interval from zero to half the decimated sample frequency. The subband may be individually processed and the processed subbands may subsequently be combined to generate a full band output signal.

Abstract: Communications equipment for use by a call center to enable communications between the call center and one or more user devices is provided, with the communications equipment comprising audio processing circuitry for processing audio signals received from the user devices over a communications network, wherein the processing circuitry is configured to perform acoustic echo cancellation and/or noise suppression and/or dereverberation processing on the received audio signals to cancel acoustic echoes and/or suppress noise and/or suppress reverberation that is present in the audio signals received from the user device.

Abstract: An audio echo suppressor includes a first receiver for receiving a first audio signal for rendering by a loud-speaker and a second receiver for receiving a microphone signal. A linear echo-cancellation filter generates a first compensation signal from the first audio signal and a compensator generates a residual signal by compensating the microphone signal for the first compensation signal. A first adapter determines filter parameters for the linear echo-cancellation filter. An estimator generates distortion measures where each distortion measure is indicative of the contribution to the residual signal in a frequency interval outside a first frequency interval. The residual signal results from rendering of signal components of the first audio signal that are within the first frequency interval. An echo reducer performs echo suppression based on the distortion measures.

Abstract: An inhaler with a housing (H) comprising an air-inlet (A_I) and a an air-outlet (A_O). Inside the housing (H) a flow path (FP) is defined between air-inlet (A_I) and air-outlet (A_O) where a dispenser (DP) is arranged to dispense an aerosol or a dry powder in the flow path (FP). Two sensors (S1, S2), e.g. microphones, are positioned spaced apart at external surfaces of the housing (H) to sense sound or vibrations resulting from a flow in the flow path (FP) at two different positions. This allows a precise detection of flow velocity during inhalation based on the sound or vibrations sensed by the two sensors (S1, S2), thus allowing examination of correct use of the inhaler. Further, the use of two spaced apart sensors (S1, S2) facilitates identification of priming and firing events in the sensed sound or vibrations which may also be used in evaluating the use of the inhaler.

Abstract: An echo cancellation device including a first adaptive filter for producing a first echo cancellation signal, a second adaptive filter for producing a second echo cancellation signal, and a post-processor for suppressing any remaining echo. The first adaptive filter and the second adaptive filter are designed for canceling a first part of the echo impulse response and a second part of the echo impulse response respectively. The device may be utilized in a mobile telephone.

Abstract: An audio echo suppressor comprises a first receiver (201) which receives a first audio signal for rendering by a loud-speaker (205) and a second receiver (211) which receives a microphone signal. A linear echo-cancellation filter (213) generates a first compensation signal from the first audio signal and a compensator (215) generates a residual signal by compensating the microphone signal for the first compensation signal. A first adapter (217) determines filter parameters for the linear echo-cancellation filter (213). An estimator (219) generates distortion measures where each distortion measure is indicative of the contribution to the residual signal in a frequency interval outside a first frequency interval. The residual signal results from rendering of signal components of the first audio signal that are within the first frequency interval. An echo reducer (207) then performs echo suppression based on the distortion measures.

Abstract: A processing apparatus for processing a physiological signal is presented. The processing apparatus is configured to perform the steps of: obtaining the physiological signal (6) containing at least two wave cycles originating from a physiological process (6b), obtaining a feature signal (12) descriptive of occurrences of a feature (P) in the physiological signal (6), determining an average waveform of the physiological signal (6) around said occurrences of the feature (P) in the physiological signal (6), and determining a model signal (18) comprising amplitude-scaled instances of the average waveform of the physiological signal (6) placed at the occurrences of the feature (P) in the physiological signal (6). Furthermore, a corresponding processing method, a system (1) for determining a vital sign of a subject comprising said processing apparatus and a computer program are presented.

Abstract: An improved determination of an acoustic coupling between a loudspeaker and a microphone, which results in a better estimate of the acoustic coupling during double talk, is achieved in a device for determining an acoustic coupling between a far-end talker signal as produced by the loudspeaker and a combined signal as picked up by the microphone. The combined signal comprises an echo of the far-end talker signal and a near-end talker signal. The acoustic coupling is derived from a ratio of changes in an envelope of the combined signal to changes in the envelope of the far-end talker signal in a predetermined time interval. In this way, a systematic error during double talk is strongly reduced.

Abstract: A multi-channel acoustic echo canceller arrangement comprises a microphone (111) providing a microphone signal having contributions from at least two audio sources (107, 109) to be cancelled. An echo canceling circuit (113, 115) performs echo cancellation of the two audio sources (107, 109) based on channel estimates for channels from each of the audio sources (107, 109) to the microphone (111). An estimation circuit (117) generates each of the channel estimates as a combination of a previous channel estimate and a channel estimate update where the combination includes applying a relative weight to the channel estimate update relative to the previous channel estimate. A weight processor 119 varies the relative weight in response to a time value. The arrangement may provide improved echo-cancellation for scenarios wherein the rendering of sound from the audio sources (107, 109) is time varying, such as when time varying decorrelation filters are used.