74 results about "Amplitude scaling" patented technology

The invention relates to a multiple-template matching identity recognition method based on an ECG (Electrocardiogram) under an electrocardiogram abnormality state, and belongs to the technical field of biological characteristic identity recognition. The ECG data of a user to be recognized is compared with the data of a registered user in a template library to obtain an identity recognition result. The key technology of the method comprises the following steps: carrying out electrocardiosignal preprocessing for eliminating noise interference; carrying out electrocardiosignal decomposition to separate an electrocardiogram waveform of each period; carrying out standardized processing for independently achieving standardization on time and amplitude scales; carrying out characteristic extraction: in the step, characteristics are extracted by wavelet transform, and clustering analysis is carried out by an ISODATA (Iterative Self-organizing Data Analysis Techniques Algorithm) so as to construct an ECC template library; and carrying out correlation analysis: in the step, correlation between ECG test data and each template is calculated, an optimal matching template is selected, and finally, an identity recognition result is obtained. The multiple-template matching identity recognition method provided by the invention utilizes the intrinsic electrocardiosignal of a human body to recognize an identity, and the ECG data under the abnormality state is considered.

Disclosed is an amplitude-scaling resilient audio watermarking apparatus and method. An encoding apparatus includes: a polyphase filterbank for dividing an audio signal into a plurality of subbands; a psychoacoustic module for applying a psychoacoustic model to the audio signal to provide a signal-to-mask ratio; a watermark encoder for evaluating an encoding parameter from the subbands according to the signal-to-mask ratio and embedding the encoding parameter and a watermark into subbands corresponding to the middle frequency; and a synthesis filterbank for synthesizing the subband signals to output a watermarked audio signal. A decoding apparatus includes: a polyphase filterbank for dividing a received audio signal into the predetermined number of subbands; an EM estimator for estimating an scale factor from an encoding parameter contained in the audio signal and a watermarked subband according to an EM algorithm, and generating the quantizer step size Δ_d of a decoder according to the amplitude-scaling; a watermark decoder for extracting a watermark from a subband corresponding to the middle frequency considering the quantizer step size; and an integrated determiner for integrating outputs of the watermark decoder to determine a watermark.

Method and system for channel-invariant robust audio hashing, the method comprising:

• • a robust hash extraction step wherein a robust hash is extracted from audio content, said step comprising:
    • dividing the audio content in frames;
    • applying a transformation procedure on said frames to compute, for each frame, transformed coefficients;
    • applying a normalization procedure on the transformed coefficients to obtain normalized coefficients, wherein said normalization procedure comprises computing the product of the sign of each coefficient of said transformed coefficients by an amplitude-scaling-invariant function of any combination of said transformed coefficients;
    • applying a quantization procedure on said normalized coefficients to obtain the robust hash of the audio content; and
  • a comparison step wherein the robust hash is compared with reference hashes to find a match.

The invention relates to a multi-wave AVO reservoir elastic parameter inversion method based on the precise Zoeppritz equation. The method includes: a. defining amplitude scaling factors; b. counting the prior information of model parameters and calculating the covariance matrix of the relevance of the three parameters; c. building an initial elastic parameter model of the time domain; d. forward modeling the multi-wave seismic pre-stack gather based on the elastic parameter mode and the precise Zoeppritz equation and calculating the inversion residual error between the PP wave and the PS wave according to the forward modeling records and the actual records; e. building and solving the inversion objective function and obtaining the solution expression of the disturbance quantity of the model parameters; f. calculating the disturbance quantity of the model parameters according to the inversion residual error and the solution expression of the disturbance quantity; g. repeating and iterating the steps of d, e and f and obtaining an optimal model parameter. The multi-wave AVO reservoir elastic parameter inversion method based on the precise Zoeppritz equation effectively improves the calculating precision and the stability of the existing pre-stack AVO inversion method, and meets the requirement of the inversion identification of oil and gas reservoir of the seismic prestack, especially the requirement of the characterization of the shale gas reservoir.

A system and computer based method for transcribing and extracting metadata from a source media. A processor-based server extracts audio and video stream from the source media. A speech recognition engine processes the audio and/or video stream to transcribe the audio and/or video stream into a time-aligned textual transcription and to extract audio amplitude by time interval, thereby providing a time-aligned machine transcribed media. The server processor measures the aural amplitude of the extracted audio amplitude and assigns a numerical value that is normalized to a single, normalized, universal amplitude scale. A database stores the time-aligned machine transcribed media, time-aligned video frames and the assigned value from the normalized amplitude scale.

Methods and devices for determining the position and/or angular orientation of a rotating shaft. Exemplary features include a sensor module and a position determination module. Sensor module may include a plurality of Hall Effect devices (HEDs) arranged at a specified angular separation to produce a signal in response to rotation of the shaft. Position module may be responsive to sensor module to produce a converted signal, determine an error term, and produce a position estimate. Converted signal may be produced by processing the HED signals into sinusoidal reference signals having offset scale and amplitude scale factors. Error term may be determined by processing the converted signals to produce an estimated position signal. Position estimate may be produced by processing the error term. Refined position measurement may be achieved by iterative elimination of regressive differences between position estimates with minimization of absolute magnitude of error term.

Multicarrier modulated communications, involving a transmitting modem (30; 30′) and a receiving modem (40; 40′) that operate according to normal operating mode and a low-power, quiescent, mode, are disclosed. In the quiescent mode, transmitter power is saved by scaling down the amplitude of the transmitted symbols in each frame by a scaling factor (A), and transmitting multiple instances of the frames in sequence. On receipt, the repeated frames are summed, and the summed amplitudes scaled to account for the repetition factor (M) and the scaling factor (A), to recover the transmitted information. If the repetition factor (M) is greater than or equal to the square of the scaling factor (A), the signal-to-noise ratio in quiescent mode is at least as high as that in normal operation. The repeating, scaling, and summing may all be performed in the frequency domain or in the time domain, as desired.

Analog signals can be fully encoded as an asynchronous time sequence generated by a time encoding machine. With knowledge of the parameters of the time encoding machine, the asynchronous time sequence can be decoded using a non-linear time decoding machine. Such a system can be extended into an encoder/decoder in which a signal is processed in M separate channels. An input signal is applied to the encoder where the signal is provided to an M channel encoder circuit including a filter bank having a total bandwidth partitioned among M adjacent or overlapping filters. Each of the M filters are coupled to a corresponding one of M time encoding machines. The encoder output is represented by M sets of time encoded trigger values. The input signal can be recovered from the M sets of time encoded trigger values by applying the trigger signals to a corresponding M channel decoder which includes M TDMs and filters. The TDMs recover the continuous signal from each channel. The filter outputs xm are then amplitude scaled sm and are combined to recover the input signal. By partitioning the signal bandwidth into M channels, the average pulse rate from each TDM is substantially reduced.

The invention belongs to the technical field of data acquisition, and more particularly relates to a method for extending a dynamic range of an analog-digital converter (ADC) in a data acquisition system. The method for extending the dynamic range of the data acquisition system comprises following steps of equally dividing an analog signal received by the data acquisition system into n paths of parallel signals, wherein n is greater than or equal to 2; performing unequal amplitude scaling on each path of parallel signal, and then connecting each path of parallel signal with an ADC; and recovering data parallelly sampled and quantified by a plurality of paths of ADCs into single path of sampling signal through corresponding amplitude recovery and data dynamic reconstruction algorithm in a digital domain. According to the method, fewer ADCs are adopted, and the amplitude range of the quantifiable signal is extended; in addition, quantification accuracy of small signals is guaranteed, both high sampling rate and high resolution of a single ADC are guaranteed, and the dynamic range of the data acquisition system is extended.

The present invention provides a method and associated apparatus providing a signal metric for a transmitter of a communication system. While the transmitter transmitting a signal with amplitudes scaled by scaling combinations selected from a plurality of scaling combination alternatives, the method including: providing a plurality of power terms, each power term corresponding to one of the scaling combination alternatives; and calculating the signal metric according to the plurality of power terms.

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.

The invention relates to an improved dither modulation (DM) watermarking method and belongs to the technical field of digital watermarks. Compared with a statistical watermarking method of known carrier signal such as spread spectrum watermarking, original DM has the advantages of higher watermark capacity and has the disadvantage of high sensitivity to amplitude scaling attack. The invention constructs a signal invariable relative to the scaling for quantification by using a method which comprises: extracting from original content a statistical characteristic sequence which has a characteristic of scaling with original data according to the same scale factor; and obtaining the signal invariable relative to the amplitude scaling by dividing a carrier signal by the introduced characteristic sequence correspondingly. Watermark embedment and detection operation are both carried out on the newly constructed signal. The method improves the robustness of watermarks to amplitude scaling greatly and obtains watermarking performance similar to the original STDM watermarking in other aspects.

The invention discloses a method for detecting the quality of a physiological signal based on a self-correlation function. According to the method disclosed by the invention, a physiological signal model under a theoretical condition is self-correlated with a physiological signal acquired under a practical condition; a relationship between the theoretical condition and the practical condition is obtained by analysis; simultaneously, the invention provides a universal method for detecting the quality of the physiological signal by utilizing the characteristic that the physiological signal has quasi-periodicity; and the method comprises the steps of respectively obtaining amplitude scaling factors under the theoretical condition and the practical condition, judging whether a difference value rate satisfies set requirements or not, and obtaining a quality detecting result. The method disclosed by the invention is applicable to all physiological signals; furthermore, the quality detecting result can be obtained only by adopting a certain segment of function in the physiological signals; therefore, the calculation amount is effectively reduced; and the signals can be evaluated in real time.

A self-calibrating human blood pressure measurement method provided by the present invention comprises the following steps: collecting human blood pressure signals, and performing ADC sampling on the collected human blood pressure signals; performing system error elimination processing on the ADC sampling data; sampling according to the elimination of system errors The pulse wave envelope is formed by data fitting, and the average blood pressure of the human body and the inflection points on both sides of the peak of the pulse wave envelope are determined through the pulse wave envelope; The proportional coefficient between the amplitude and the amplitude of the pulse wave envelope corresponding to the average blood pressure of the human body; judge the position of the inflection point on both sides of the peak of the pulse wave envelope line, and according to the position of the inflection point and the pulse wave envelope corresponding to the systolic blood pressure and the diastolic blood pressure of the human body The proportional relationship between the amplitude of the curve and the amplitude of the pulse wave envelope corresponding to the average blood pressure of the human body determines the systolic blood pressure and the diastolic blood pressure of the human body; it can effectively eliminate the systematic error in the blood pressure sampling data; it reduces the difficulty of hardware design and the complexity of operation Degree, reduce product cost.

The invention relates to a method for identifying environment sound events based on time spectrum amplitude scaling vectors. Firstly, a time spectrum amplitude scale vector (TSASV) of each related sound event is computed as an identifying prototype, and is stored in a database to be used as a template for identifying a sound event to be detected; and then the TSASV of the sound event to be detected is computed and is compared with each identifying prototype, and the sound event proximate to the identifying prototype is the sound event to be identified. The construction method of the TSASV comprises the steps that fast Fourier transform is carried out on the obtained environment sound event to generate a spectrogram; the generated spectrogram is sampled according to different frequency resolutions, and sub spectrograms with different frequency resolutions are constructed by the spectrogram; signal enhancement is carried out on the sub spectrograms to generate signal-enhanced sub spectrograms; amplitude scaling is carried out on the signal-enhanced sub spectrograms; and the sub spectrograms after amplitude scaling are coded to generate the TSASV. The method is conductive to improving the effectiveness of environment sound event identification.

A method of quadrature amplitude modulation involving encoding phase differentially and amplitude absolutely, allowing for a high data rate and spectral efficiency in data transmission and other communication applications, and allowing for amplitude scaling to facilitate data recovery; amplitude scale tracking to track-out rapid and severe scale variations and facilitate successful demodulation and data retrieval; 2<N >power carrier recovery; incoherent demodulation where coherent carrier recovery is not possible or practical due to signal degradation; coherent demodulation; multipath equalization to equalize frequency dependent multipath; and demodulation filtering.

The invention belongs to the technical field of the design of the main force-bearing component of an aerospace structure, and relates to a three-dimensional finite element model trimming method capable of embodying manufacture geometric defects. The method comprises the following steps of: 1) adopting an optical measurement method to carry out measurement to obtain three-dimensional point cloud information; 2) carrying out filtering processing on the three-dimensional point cloud information to eliminate noise; 3) establishing a perfect structure three-dimensional finite element model which does not consider the three-dimensional point cloud information; and 4) on the basis of the three-dimensional point cloud information, or after the obtained three-dimensional point cloud information is subjected to amplification and reduction processing, adopting an interpolation method to trim the perfect structure three-dimensional finite element model, and embodying the manufacture geometric defects. By use of the method, the authenticity of a finite element prediction model can be enhanced, furthermore, the accuracy of finite element prediction can be improved, meanwhile, an existing finite element model can be guaranteed to be fully utilized to realize adaptive defect amplitude scaling and position selection, a workload is extremely low, and the method is expected to become one of the key technologies of finite element model correction in an aerospace field.

A clipping circuit to limit amplitude of an orthogonal baseband signal includes a 16-gonal clipping circuit almost equivalent to a circular clipping circuit. The 16-gonal clipping circuit includes plural stages of series-connected units each including a rectangular clipping circuit and a phase rotating module. To compensate for variation in the signal amplitude due to the rectangular clipping circuits and the phase rotating modules, an amplitude adjuster conducts amplitude scaling for the signal.

The invention provides a precise machine stand vibration attenuation base capable of measuring the amplitude. The precise machine stand vibration attenuation base comprises an upper vibration attenuation block, a lower vibration attenuation block and a plurality of pin rollers; the upper vibration attenuation block is fixedly connected with a precise machine stand, and the multiple pin rollers are arranged between the lower surface of the upper vibration attenuation block and the upper surface of the lower vibration attenuation block in a tightly-pressed manner; the lower surface of the upper vibration attenuation block is an outwards-convex arc face, and the upper surface of the lower vibration attenuation block is an inwards-concave arc face matched with the outwards-convex arc face; and an amplitude scale strip is arranged on one lateral face of the lower vibration attenuation block, and a marking strip used for pointing to the amplitude scale strip is arranged on the corresponding lateral face of the upper vibration attenuation block. According to the precise machine stand vibration attenuation base, as the structures of pointer scales and the pin rollers are ingeniously combined, the influences of vibration on the precise machine stand can be reduced, meanwhile, the amplitude can be measured, and data supports are further provided for vibration attenuation. The precise machine stand vibration attenuation base is simple in structure and reasonable in design.

A receiver apparatus and method are provided blind amplitude estimation for a received complex signal, wherein the complex signal is analyzed to determine first and second order statistics of real part and imaginary part of the complex signal. Based on a predetermined non-linear function, amplitude scaling information is pre-calculated and used to derive a desired amplitude scaling information for said determined first and second order statistics. Thereby, a low complexity method for blind amplitude estimation for unknown data symbols distorted by an unknown amount of noise can be provided.