The present invention relates to a method of determining the magnetic flux φ of an electric machine, based on measurements (MES) of currents and voltages in the phases of the electric machine, on a dynamic model (MOD) of the magnetic flux and on an adaptive Kalman filter (KAL).

In order to reduce the amount of calculation for signal distortion compensation, this filter coefficient updating device for updating the filter coefficients of a plurality of filters in a filter layer comprising the plurality of filters, which are connected in a first plurality of stages with respect to received data, is provided with: a deriving unit for deriving the respective filter coefficients of the plurality of filters in one or a plurality of stages included in the first plurality of stages, by means of output data output from the last stage of the first plurality of stages; and an updating unit for updating each of the filter coefficients.

The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing a multimedia signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing filtering for multimedia signal having a plurality of subbands with a low calculation amount. To this end, provided are a method for processing a multimedia signal including: receiving a multimedia signal having a plurality of subbands; receiving at least one proto-type filter coefficients for filtering each subband signal of the multimedia signal; converting the proto-type filter coefficients into a plurality of subband filter coefficients; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the length of at least one truncated subband filter coefficients being different from the length of truncated subband filter coefficients of another subband; and filtering the multimedia signal by using the truncated subband filter coefficients corresponding to each subband signal and an apparatus for processing a multimedia signal using the same.

An embodiment provides a closed loop active gate driver configured to drive a switch for an inductive load and including a feedback loop, the feedback loop configured to sample a repetitive output waveform of the inductive load, the output waveform having a plurality of repetitive cycles and the feedback loop configured to sample the output waveform using a sampling rate that is lower than a sampling rate required for characterizing the output waveform, sample points acquired in cycles of the plurality of cycles are acquired at different time points during the cycles and wherein a representation of the output waveform is reconstructed using the sample points.

Examples provide a method and apparatus for an analog bi-quad infinite impulse response (IIR) filter. An amplifier generates a positive output signal corresponding to a received RF signal and a negative output signal. A set of selectively switchable time-delay circuits associated with a positive arm of the filter causes a predetermined delay corresponding to a desired sample frequency. A first set of configurable variable gain amplifiers amplify the positive output signal to establish a set of positive coefficients. A set of selectively switchable time-delay circuits associated with a negative arm of the filter causes a predetermined delay. A delayed negative output signal is generated which is amplified by a second set of configurable variable gain amplifiers to establish a set of negative coefficients. A set of power combiners function as sum junctions to combine the delayed positive output signals and the delayed negative output signals into a single output signal.

An electronic device, an echo signal cancelling method thereof and a non-transitory computer readable recording medium is provided. The electronic device according to an exemplary embodiment includes a speaker configured to output a sound corresponding to a reference signal, a microphone configured to generate a microphone signal by obtaining a received sound, and a filter configured to cancel an echo signal of the reference signal from the microphone signal. In addition, the filter includes a first filter configured to estimate an echo signal of the reference signal and cancel the estimated echo signal from the microphone signal, and a second filter configured to generate an adaptive gain to cancel a residual echo from the microphone signal in which the estimated echo signal is canceled, and generate an output signal by using the generated adaptive gain and the microphone signal in which the estimated echo signal is canceled.

The adaptive filter for sparse system identification is an adaptive filter that uses an algorithm in the feedback loop that is designed to provide better performance when the unknown system model is sparse, i.e., when the filter has only a few non-zero coefficients, such as digital TV transmission channels and echo paths. The algorithm is a least mean square algorithm with filter coefficients updated at each iteration, as well as a step size that is also updated at each iteration. The adaptive filter may be implemented on a digital signal processor (DSP), an application-specific integrated circuit (ASIC), or by field-programmable gate arrays (FPGAs).

A method is disclosed for audio object extraction from an audio content which includes identifying a first set of projection spaces including a first subset for a first channel and a second subset for a second channel of the plurality of channels. The method may further include determining a first set of correlations between the first and second channels, each of the first set of correlations corresponding to one of the first subset of projection spaces and one of the second subset of projection spaces. Still further, the method may include extracting an audio object from an audio signal of the first channel at least in part based on a first correlation among the first set of correlations and the projection space from the first subset corresponding to the first correlation, the first correlation being greater than a first predefined threshold. Corresponding system and computer program products are also disclosed.

A method is disclosed for audio object extraction from an audio content which includes identifying a first set of projection spaces including a first subset for a first channel and a second subset for a second channel of the plurality of channels. The method may further include determining a first set of correlations between the first and second channels, each of the first set of correlations corresponding to one of the first subset of projection spaces and one of the second subset of projection spaces. Still further, the method may include extracting an audio object from an audio signal of the first channel at least in part based on a first correlation among the first set of correlations and the projection space from the first subset corresponding to the first correlation, the first correlation being greater than a first predefined threshold. Corresponding system and computer program products are also disclosed.

A variable update step size is determined in proportion to a magnitude ratio or magnitude difference between a first residual signal and a second residual signal. The first residual signal is obtained by using adaptive filter coefficient sequence, where the adaptive filter coefficient sequence has been obtained in previous operations of the adaptive equalizer. The second residual signal is obtained by using a prior update adaptive filter coefficient sequence, where the prior update adaptive filter coefficient sequence is obtained by performing a coefficient update with an arbitrary prior update step size on the adaptive filter coefficient sequence having been obtained in previous operations of the adaptive equalizer.