Abstract: The present invention provides a method of identifying a subject for whom a prostate biopsy is indicated, comprising: a) determining, from a nucleic acid sample obtained from the subject, a genotype for the subject at a plurality of biallelic polymorphic loci, wherein each of said plurality has an associated allele and an unassociated allele, wherein the genotype is selected from the group consisting of homozygous for the associated allele, heterozygous, and homozygous for the unassociated allele; b) calculating a genetic risk score (GRS) for the subject based on the genotype determined in step (a); and c) analyzing the GRS of the subject in combination with a prostate specific antigen (PSA) level of the subject to identify a prostate cancer detection rate for the subject, whereby a prostate cancer detection rate of greater than or equal to a reference value identifies the subject as a subject for whom a prostate biopsy is indicated.

Abstract: The invention relates to a method for determining an encoding parameter for an audio channel signal of a multi-channel audio signal, the method comprising: determining for the audio channel signal a set of functions from the audio channel signal and a reference audio signal; determining a first set of encoding parameters based on a smoothing of the set of functions with respect to a frame sequence of the multi-channel audio signal, the smoothing being based on a first smoothing coefficient; determining a second set of encoding parameters based on a smoothing of the set of functions with respect to the frame sequence of the multi-channel audio signal, the smoothing being based on a second smoothing coefficient; and determining the encoding parameter based on a quality criterion with respect to the first set of encoding parameters and/or the second set of encoding parameters.

Abstract: Described are a read-write control method for memory, and a corresponding memory and server. The method comprises: dividing a storage resource of the memory by taking a block as a unit; to write data, combining the data to be written into a data block, writing the data block in a free segment of the memory, and recording an identifier and a corresponding index of the data, the index of the written data is an offset indicating a memory location of the data in the memory; to read data, reading the data from the offset of the memory according to an identifier and a index of the data to be read. When read-write control is performed on the memory, the number of the times that the I/O operation is performed on the memory is reduced during data writing, thereby improving the efficiency of the write operation on the memory.

Abstract: A video content generation device generates video data synchronized with music data based on motion data, representing a motion graph including nodes, edges, and weights, and metadata indicating a synchronization probability per each node between the motion graph and a musical tune. A music data storage unit stores the predetermined amount of music data and their musical characteristics in connection with the predetermined number of beats, in a reproduction order, retrieved from the musical tune. An optimum path search unit searches an optimum path connecting between nodes, each of which is selected per each beat with a high synchronization probability, on the motion graph with motion characteristics matching with musical characteristics based on the predetermined amount of music data. Video data synchronized with music data is generated based on synchronization information for correlating motion data to music data along with the optimum path.

Abstract: A framing method and apparatus are disclosed to overcome inconsistency of gains between sub-frames caused by simple average framing in the prior art. The method includes: obtaining the Linear Prediction Coding (LPC) order and the pitch of the signal; removing the samples inapplicable to Long-Term Prediction (LTP) synthesis according to the LPC prediction order and the pitch; and splitting the remaining samples of the signal into several sub-frames. The technical solution under the present invention is applicable to the multimedia speech coding field.

Abstract: An apparatus and a method for encoding an input signal applied to the apparatus comprising: a transient detector adapted to detect whether the applied input signal comprises a transient; at least two transient signal encoders adapted to encode the applied input signal if a transient is detected by the transient detector; and a selection unit adapted to select a transient signal encoder among the transient signal encoders according to at least one predetermined selection criterion.

Abstract: The present disclosure relates to coding and decoding technologies, and discloses a preprocessing method, a preprocessing apparatus, and a coding device. The preprocessing method includes: obtaining characteristic information of a current frame signal; identifying whether the current frame signal requires no coding operation of removing LTC according to the characteristic information of the current frame signal and preset information; and if identifying that the current frame signal requires no coding operation of removing LTC, performing the coding operation of removing STC for the current frame signal; and if identifying that the current frame signal requires the coding operation of removing LTC, performing the coding operations of removing both LTC and STC for the current frame signal. Through the technical solution provided herein, the coding operation of removing LTC is performed for only part of the input frame signals.

Abstract: The present invention relates to communication technologies and discloses a method, an apparatus and a system for Linear Prediction Coding (LPC) analysis to improve LPC prediction performance and simplify analysis operation. The method includes: obtaining signal feature information of at least one sample point of input signals; comparing and analyzing the signal feature information to obtain an analysis result; selecting a window function according to the analysis result to perform adaptive windowing for the input signals and obtain windowed signals; and processing the windowed signals to obtain an LPC coefficient for linear prediction. The embodiments of the present invention are applicable to LPC.

Abstract: A voice or audio signal processor for processing received network packets received over a communication network to provide an output signal, the voice or audio signal processor comprising a jitter buffer being configured to buffer the received network packets, a voice or audio decoder being configured to decode the received network packets as buffered by the jitter buffer to obtain a decoded voice or audio signal, a controllable time scaler being configured to amend a length of the decoded voice or audio signal to obtain a time scaled voice or audio signal as the output voice or audio signal, and an adaptation control means being configured to control an operation of the time scaler in dependency on a processing complexity measure.

Abstract: Methods and devices for a low complex inter-channel phase difference estimation are provided. A method for the estimation of inter-channel phase differences (IPDs), comprises applying a transformation from a time domain to a frequency domain to a plurality of audio channel signals, calculating a plurality of IPD values for the IPDs between at least one of the plurality of audio channel signals and a reference audio channel signal over a predetermined frequency range, each IPD value being calculated over a portion of the predetermined frequency range, calculating, for each of the plurality of IPD values, a weighted IPD value by multiplying each of the plurality of IPD values with a corresponding frequency-dependent weighting factor, and calculating an IPD range value for the predetermined frequency range by adding the plurality of weighted IPD values.

Abstract: The present disclosure relates to a method, apparatus, and system for encoding and decoding signals. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag value according to a decision criterion; obtaining a second-domain predictive signal according to the LP processing result and the LTP processing result when the long-term flag value is a first value; obtaining a second-domain predictive signal according to the LP processing result when the long-term flag value is a second value; converting the second-domain predictive signal into a first-domain predictive signal, and calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal.

Abstract: A method for parametric spatial audio coding of a multi-channel audio signal comprising a plurality of audio channel signals is provided, the method comprising: calculating at least two different spatial coding parameters for an audio channel signal of the plurality of audio channel signals, selecting at least one spatial coding parameter of the at least two different spatial coding parameters associated with the audio channel signal on the basis of the values of the calculated spatial coding parameters; including a quantized representation of the selected spatial coding parameter into a parameter section of an audio bitstream; and setting a parameter type flag in the parameter section of the audio bitstream indicating the type of the selected spatial coding parameter being included into the audio bitstream.

Abstract: The invention relates to a parametric audio encoder, comprising a parameter generator, the parameter generator being configured to determine a first set of encoding parameters and reference audio signal values, wherein the reference audio signal is another audio channel signal or a downmix audio signal derived from at least two audio channel signals of the plurality of multi-channel audio signals, to determine a first encoding parameter average based on the first set of encoding parameters of the audio channel signal, to determine a second encoding parameter average based on the first encoding parameter average of the audio channel signal and at least one other first encoding parameter average of the audio channel signal, and to determine the encoding parameter based on the first encoding parameter average of the audio channel signal and the second encoding parameter average of the audio channel signal.

Abstract: The present invention provides methods of assessing an individual subject's risk of developing prostate cancer, comprising: a) analyzing a nucleic acid sample obtained from the subject and determining a genotype for the subject at a plurality of biallelic polymorphic loci, wherein each of said plurality has an associated allele and an unassociated allele, wherein the genotype is selected from the group consisting of homozygous for the associated allele, heterozygous, and homozygous for the unassociated allele; and b) calculating a cumulative relative risk (CRR) for the subject based on the genotype determined in step (a). A CRR of greater than 1.00 identifies a subject as having an increased risk of developing prostate cancer and also can identify a subject who is a candidate for early PSA screening, prostate biopsy and/or chemoprevention.

Abstract: Methods of increasing the yield in plant expression of recombinant proteins comprising engineering glycosylation sites into cloned genes or cDNAs for proteins using codons that drive post-translational modifications in plants; and engineering the cloned genes or cDNAs to contain a plant secretory signal sequence that targets the gene products (protein) for secretion. The methods result in increased recombinant glycosylated protein yields. Proteins produced according to these methods are disclosed.

Abstract: A method and apparatus for providing signal processing coefficients for processing an input signal at a predetermined signal processing sampling rate, wherein the input signal is received at an input signal sampling rate, the method comprising the steps of computing a correlation or covariance function based on the received input signal at the input signal sampling rate to provide correlation or covariance coefficients at the input signal sampling rate, re-sampling the computed correlation or covariance coefficients having the input signal sampling rate to provide correlation or covariance coefficients at the predetermined signal processing sampling rate, and calculating the signal processing coefficients based on the correlation or covariance coefficients at the predetermined signal processing sampling rate.

Abstract: The present disclosure relates to a method, apparatus, and system for encoding and decoding signals. The encoding method includes: converting a first-domain signal into a second-domain signal; performing Linear Prediction (LP) processing and Long-Term Prediction (LTP) processing for the second-domain signal; obtaining a long-term flag value according to a decision criterion; obtaining a second-domain predictive signal according to the LP processing result and the LTP processing result when the long-term flag value is a first value; obtaining a second-domain predictive signal according to the LP processing result when the long-term flag value is a second value; converting the second-domain predictive signal into a first-domain predictive signal, and calculating a first-domain predictive residual signal; and outputting a bit stream that includes the first-domain predictive residual signal.

Abstract: A voice quality enhancement (VQE) detector for a network element receiving an audio signal from a previous network element of a network, wherein said voice quality enhancement detector is adapted: to perform a voice quality enhancement detection based on the received audio signal, wherein said voice quality enhancement detection comprises detecting that at least one voice quality enhancement function, VQEF, was applied to the received audio signal by at least one previous network element of the network; and to control a voice quality enhancement processing of the received audio signal depending on the detection result.

Abstract: A data processing method is disclosed, including: twiddling input data, so as to obtain twiddled data; pre-rotating the twiddled data by using a symmetric rotate factor, where the rotate factor is a·W4L2p+1, p=0, . . . , L/2?1, and ? is a constant; performing a Fast Fourier (Fast Fourier Transform, FFT) transform of L/2 point on the pre-rotated data, where L is the length of the input data; post-rotating the data that has undergone the FFT transform by using a symmetric rotate factor, where the rotate factor is b·W4L2q+1, q=0, . . . , L/2?1, and b is a constant; and obtaining output data.

Abstract: The present disclosure relates to coding and decoding technologies, and discloses a preprocessing method, a preprocessing apparatus, and a coding device. The preprocessing method includes: obtaining characteristic information of a current frame signal; identifying whether the current frame signal requires no coding operation of removing LTC according to the characteristic information of the current frame signal and preset information; and if identifying that the current frame signal requires no coding operation of removing LTC, performing the coding operation of removing STC for the current frame signal; and if identifying that the current frame signal requires the coding operation of removing LTC, performing the coding operations of removing both LTC and STC for the current frame signal. Through the technical solution provided herein, the coding operation of removing LTC is performed for only part of the input frame signals.