Abstract: According to an aspect, there is provided a computer-implemented method for determining an arterial blood oxygenation, SpO2, measurement for a subject. The method comprises receiving (101) a first and second photoplethysmogram, PPG, signals for the subject. The first PPG signal is obtained using a first wavelength of light and the second PPG signal is obtained using a second wavelength of light, and the first and second PPG signals comprise pulse components relating to blood volume changes in the subject over time. The method also comprises determining (103) a fundamental frequency corresponding to a pulse rate of the subject occurring during the measurement of the first and second PPG signals, and processing (105) one or more higher harmonics of the pulse components of the first and second PPG signals at frequencies higher than the fundamental frequency, excluding the fundamental frequency of the pulse components, to determine an SpO2 measurement for the subject.

Abstract: A system (8) and method is for extracting from sensed induction signals, component signals pertaining to different physiological phenomena in the body. A resonator circuit (10) is oscillated at a certain frequency to generate an alternating electromagnetic field which is applied to a body to be investigated. This field induces secondary eddy currents in the body which interact with the primary magnetic field and alter at least the frequency and amplitude of the resonator circuit oscillating current. These changes in the current characteristics, in particular the frequency and amplitude, are measured and provide first and second input signals. A system (8) or method is provided by embodiments of the invention which is arranged to receive these input signals. A multitude of different composite or fused signals are then generated by the system, each formed from a different linear combination ratio of the two input signals.

Abstract: An inductive sensing system (8) is adapted to apply electromagnetic excitation signals into a body, the system comprising a resonator circuit (10) incorporating a loop antenna (12). The system senses signals returned back from the body with the same antenna, based on variation in electrical characteristics of the resonator circuit. The system is configured for separating signals received from different physiological sources within the body. This is performed based on detecting in the resonator circuit electrical characteristics indicative of both a real and an imaginary part of an additional inductance component added to the antenna by received electromagnetic signals. The separating the signals from different physiological sources is based on relative magnitudes of said detected real and imaginary inductance components added to the resonator circuit by the returned signals.

Abstract: According to an aspect, there is provided a method of determining the respiration rate of a subject, the method comprising obtaining a signal from a sensor that is worn or carried by the subject; analyzing the signal to determine a plurality of values for a breathing-related feature; forming a histogram from the plurality of values for the breathing-related feature, the histogram comprising a plurality of groups, with each group having an associated count that is the number of occurrences of a value or values for the breathing-related feature corresponding to the group; applying a weighting to the count associated with each group to form weighted counts; and determining the respiration rate from a mean of the histogram with the weighted counts.

Abstract: According to an aspect, there is provided a method of determining the respiration rate of a subject, the method comprising obtaining a signal from a sensor that is worn or carried by the subject; analyzing the signal to determine a plurality of values for a breathing-related feature; forming a histogram from the plurality of values for the breathing-related feature, the histogram comprising a plurality of groups, with each group having an associated count that is the number of occurrences of a value or values for the breathing-related feature corresponding to the group; applying a weighting to the count associated with each group to form weighted counts; and determining the respiration rate from a mean of the histogram with the weighted counts.

Abstract: The present invention relates to a method for estimating a pattern, in particular a pitch and/or a fundamental frequency, in a signal having a periodic, quasiperiodic or virtually periodic component, wherein the signal is transformed from a time-domain to a frequency-domain to obtain a spectrum of the signal, the spectrum is processed to obtain a zero-phase spectrum of the signal, the spectrum of the signal is transformed to the time-domain to obtain a correlation signal, the spectrum and the correlation signals are combined to a combined spectrum, and the pattern is estimated on the basis of the combined spectrum.

Abstract: The invention relates to a power consumption monitoring apparatus (11) for monitoring the power consumed in a network (1) of electrical devices (3, 4, 5; 6, 7, 8). An electrical parameter measuring unit measures an overall electrical parameter of the network over time,a group signature providing unit provides group signatures, wherein each group signature is indicative of an electrical parameter of a certain group (9; 10) of electrical devices over time, and a disaggregation unit disaggregates the measured overall electrical parameter over time for determining the power consumption of the individual groups of electrical devices depending on the provided group signatures. This allows disaggregating the overall electrical parameter, without providing for each group an individual measuring unit, thereby reducing the required hardware and the time needed for installing and maintaining the power consumption monitoring apparatus.

Abstract: The present invention relates to a method for estimating a pattern, in particular a pitch and/or a fundamental frequency, in a signal having a periodic, quasiperiodic or virtually periodic component, wherein the signal is transformed from a time-domain to a frequency-domain to obtain a spectrum of the signal, the spectrum is processed to obtain a zero-phase spectrum of the signal, the spectrum of the signal is transformed to the time-domain to obtain a correlation signal, the spectrum and the correlation signals are combined to a combined spectrum, and the pattern is estimated on the basis of the combined spectrum.

Abstract: The present invention relates to a method of synthesizing a signal comprising the steps of determining a required pitch bell locations, mapping the required pitch bell locations onto the signal to provide first pitch bell locations, randomizing the first pitch bell locations to provide second pitch bell locations, windowing the signal on the second pitch bell locations to provide a pitch bell, repeating the aforementioned steps for all required pitch bell locations and performing an overlap and add operation with respect to the pitch bells in order to synthesize the signal.

Abstract: The present invention relates to a method of synthesizing of a speech signal, comprising: —assigning of a first identifier to a first class of intervals of an original speech signal and assigning of a second identifier to a second class of intervals of the original speech signal, —windowing the original speech signal to provide a number of pitch bells, —processing the pitch bells having the first identifier assigned thereto for modifying a duration of the speech signal, —performing an overlap and add operation on the processed pitch bells.

Abstract: The present invention relates to a method of synthesizing a signal comprising the steps of a) determining of a required pitch bell location, b) mapping of the required pitch bell location onto an original signal to provide a first pitch bell location, c) randomizing the first pitch bell location to provide a second pitch bell location, d) windowing of the original signal on the second pitch bell location to provide a pitch bell, e) repeating of the steps a) to d) for all required pitch bell locations and performing an overlap and add operation with respect to the pitch bells in order to synthesize the signal.

Abstract: The present invention relates to a method for analyzing speech, the method comprising the steps of: a) inputting a speech signal, b) obtaining the first harmonic of the speech signal, c) determining the phase-difference Df between the speech signal and the first harmonic.

Abstract: The present invention relates to a method of synthesizing a signal comprising the steps of: a) determining of a required pitch bell location on the signal to be synthesized. b) mapping of the required pitch bell location onto an original signal to provide a first pitch bell location, c) randomizing the first pitch bell location to provide a second pitch bell location, d) windowing of the original signal on the second pitch bell location to provide a pitch bell, e) placing the resulting pitch bell at the required pitch bell location within the domain of the signal to be synthesized, f) repeating of the steps a) to e) for all required pitch bell locations and performing an overlap and add operation with respect to the pitch bells in order to synthesize the signal.

Abstract: The present invention relates to a method of synthesizing a first sound signal based on a second sound signal, the first sound signal having a required first fundamental frequency and the second sound signal having a second fundamental frequency, the method comprising the steps of, a) determining of required pitch bell locations in the time domain of the first sound signal, the pitch bell locations being distanced by one period of the first fundamental frequency, b) providing of pitch bells by windowing the second sound signal on pitch bell locations in the time domain of the second sound signal, the pitch bell locations being distanced by one period of the second fundamental frequency, c) randomly selecting of a pitch bell from the provided pitch bells for each of the required pitch bell locations, d) performing an overlap and add operation on the selected pitch bells for synthesizing the first signal.

Abstract: A method of synthesizing a speech signal by providing a first speech unit signal having an end interval and a second speech unit signal having a front interval, wherein at least some of the periods of the end interval are appended in inverted order at the end of the first speech unit signal in order to provide a fade-out interval, and at least some of the periods of the front interval are appended in inverted order at the beginning of the second speech unit signal to provide a fade-in interval. An overlap and add operation is performed on the end and fade-in intervals and the fade-out and front intervals.

Abstract: A method and unit for substracting quantization noise from a pulse code modulated PCM signal being segmented into frames. For achieving this it is proposed to first calculate for each frame of the PCM signal a quantization noise level Bq according to an equation having parameters including n which indicates a specific sample of the PCM signal, S*min[n] which represents the minimum quantization noise level for a specific sample value s*[n] of the PCM signal, S*max[n] which represents the maximum quantization noise level for the specific sample value s*[n] of the PCM signal, w[n] which represents a window-function and W which represents the number of samples per window. Subsequently, the quantization noise as represented by the quantization noise level Bq has to be substracted from the PCM signal, preferably with the help of a suitable background noise substracting system.

Abstract: In a speech coding system with an encoder and a decoder cooperating with said encoder, the speech encoder comprises a pre-processor and an ADPCM encoder with a quantizer and step-size adaptation means, while the speech decoder comprises an ADPCM decoder with similar step-size adaptation means as in the ADPCM encoder and with a decoder, and a post-processor. The quantizer is provided with storage means containing values for a correction factor ?(c(n)) of the step-size ?(n), said correction factor being dependent on the quantizer output signal c(n). The step-size adaptation occurs in accordance with the relation: ? ? ( n + 1 ) = ? ? ( n ) · A · { ? ? ( c ? ( n ) ) if ? ? ? ? ( c ? ( n ) ) < 1 ? ? ( c ? ( n ) ) + b ? if ? ? ? ? ? ( n ) · A · c max < ? max 1 otherwise ? .

Abstract: A speech enhancement system for the reduction of background noise comprises a time-to-frequency transformation unit to transform frames of time-domain samples of audio signals to the frequency domain, background noise reduction means to perform noise reduction in the frequency domain, and a frequency-to-time transformation unit to transform the noise reduced signals back to the time-domain. In the background noise reduction means for each frequency component a predicted background magnitude is calculated in response to the measured input magnitude from the time-to-frequency transformation unit and to the previously calculated background magnitude, whereupon for each of said frequency components the signal-to-noise ratio is calculated in response to the predicted background magnitude and to said measured input magnitude and the filter magnitude for said measured input magnitude in response to the signal-to-noise ratio.

Abstract: In a speech coding system with an encoder and a decoder cooperating with said encoder, the speech encoder comprises a pre-processor and an ADPCM encoder with a quantizer and step-size adaptation means, while the speech decoder comprises an ADPCM decoder with similar step-size adaptation means as in the ADPCM encoder and with a decoder, and a post-processor. The quantizer is provided with storage means containing values for a correction factor &agr;(c(n)) of the step-size &Dgr;(n), said correction factor being dependent on the quantizer output signal c(n).

Abstract: In a speech coding system with a speech encoder and a speech decoder cooperating with said speech encoder, the speech encoder comprises a pre-processor and an ADPCM (adaptive differential pulse code modulation) encoder with a quantizer and step-size adaptation means, while the speech decoder comprises an ADPCM decoder with similar step-size adaptation means as in the ADPCM encoder and a decoder, and a post-processor. The pre-processor is provided with phase-smearing filtering means to smooth the effect of high and/or rapid energy changes at the input of the quantizer, while the post-processor is provided with filtering means inverse to said phase-smearing filtering means.