Koninklijke Philips Corporation N.v.

U.S. Philips Corporation

Nxp B.v.

Ercan Ferit Gigi

Laurentia Johanna Huijbregts

Wouter Herman Peeters

Gerardus Johannes Nicolaas Doodeman

Cornelus Hendricus Bertus Arnoldus Van Dinther

Tim Patrick Steunebrink

Carlijn Andrea Vernooij

A system () and method is for extracting from sensed induction signals, component signals pertaining to different physiological phenomena in the body. A resonator circuit () 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 () 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. These are then assessed with a signal selection procedure to identify a best candidate signal for providing a measure or indication of a particular one or more physiological phenomena. This can be based on pre-defined selection criteria, for example relating to signal characteristics of the candidate signals.

Inductive sensing system and method

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.

Method and apparatus for determining a respiration rate of a subject

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.

Method and device for estimating a pattern in a signal

The present invention relates to a method of synthesizing a signal comprising the steps of

Method of synthesizing of an unvoiced speech signal

The present invention relates to a method of synthesizing of a speech signal, comprising: &#x2014;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, &#x2014;windowing the original speech signal to provide a number of pitch bells, &#x2014;processing the pitch bells having the first identifier assigned thereto for modifying a duration of the speech signal, &#x2014;performing an overlap and add operation on the processed pitch bells.

Method for controlling duration in speech synthesis

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.

Method for synthesizing speech

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.

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.

Method of synthesis for a steady sound signal

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.

Speech synthesis using concatenation of speech waveforms

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*[n] which represents the minimum quantization noise level for a specific sample value s*[n] of the PCM signal, S*[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. 

Method and unit for substracting quantization noise from a PCM signal

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 &#x3b1;(c(n)) of the step-size &#x394;(n), said correction factor being dependent on the quantizer output signal c(n). The step-size adaptation occurs in accordance with the relation: 

Speech coding system

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. The speech enhancement device may be applied in speech coding systems, particularly PCM coding systems.

Speech enhancement device

Successive pitch periods/frequencies are accurately determined in an audio equivalent signal. Using a suitable conventional pitch detection technique, an initial value of the pitch frequency/period is determined for so-called pitch detection segments of the audio equivalent signal. Based on the determined initial value, a refined value of the pitch frequency/period is determined. To this end, the signal is divided into a sequence of pitch refinement segments. Each pitch refinement segment is associated with at least one of the pitch detection segments. The pitch refinement segments are filtered to extract a frequency component with a frequency substantially corresponding to an initially determined pitch frequency of an associated pitch detection segment. The successive pitch periods/frequencies are determined in the filtered signal.

Refinement of pitch detection

An audio equivalent signal is coded by determining a noise value for harmonic frequencies. The noise value is determined by the change of phase of the harmonics in successive segments of the signal. The noise value for a harmonic frequency represents a contribution of a periodic component and an aperiodic component to the segment at the harmonic frequency. To this end, the pitch development of the signal is determined, and the signal is broken into segments of, e.g., one or two pitch periods wide. For each of the analysis segments an amplitude value and a phase value is determined for the harmonic frequencies. The noise value for each of the harmonics is determined by comparing the phase value for the harmonic of the segment to a corresponding phase value for at least one preceding or following segment. Each segment is coded as the amplitude value and the noise value for each of the harmonics. The method is preferably used for speech synthesis.

Speech coding based on determining a noise contribution from a phase change

An audio equivalent input signal is divided into a sequence of overlapping or adjacent signal segments. A lengthened signal is synthesized by systematically maintaining or repeating respective signal segments of the sequence of segments. Repeating non-periodic segments, such as a voiceless part of a speech signal or noise in music, results in audible artefacts. The introduced periodicity is broken by dividing a signal section originating from one non-periodic source signal segment into a second sequence of signal segments with at least one of the signal segments having a duration not equal to a duration of the source signal segment and not equal to a multiple of the duration of the source signal segment. Signal segments of the second sequence are shuffled.

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Eindhoven, Netherlands

Ercan Ferit Gigi