126 results about "Time–frequency representation" patented technology

A method (200) for improving quality of output audio (126). The method can include detecting an output acoustic signal (128) and generating a receive audio signal (134) based, at least in part, on the detected output acoustic signal. A frequency domain representation (140) of the receive audio signal can be compared to a frequency domain representation (138) of a source audio signal (124) from which the output acoustic signal is generated. At least one distortion signal (142) in the receive audio signal can be identified, and the source audio signal can be selectively equalized to reduce an amplitude of the source audio signal at a frequency that correlates to the distortion signal.

Perceptual audio codecs make use of filter banks and MDCT in order to achieve a compact representation of the audio signal, by removing redundancy and irrelevancy from the original audio signal. During quasi-stationary parts of the audio signal a high frequency resolution of the filter bank is advantageous in order to achieve a high coding gain, but this high frequency resolution is coupled to a coarse temporal resolution that becomes a problem during transient signal parts by producing audible pre-echo effects. The invention achieves improved coding / decoding quality by applying on top of the output of a first filter bank a second non-uniform filter bank, i.e. a cascaded MDCT. The inventive codec uses switching to an additional extension filter bank (or multi-resolution filter bank) in order to re-group the time-frequency representation during transient or fast changing audio signal sections. By applying a corresponding switching control, pre-echo effects are avoided and a high coding gain and a low coding delay are achieved.

A method of removing noise and interference from a signal by receiving the signal, calculating a joint time-frequency domain of the signal, estimating instantaneous frequencies of the joint time-frequency domain, modifying each estimated instantaneous frequency, if necessary, to correspond to a frequency of the joint time-frequency domain to which it most closely compares, redistributing the elements within the joint time-frequency domain according to the estimated instantaneous frequencies as modified, computing a magnitude for each element in the joint time-frequency domain as redistributed, plotting the results as the time-frequency representation of the signal, identifying in the plot any noise and interference components in the received signal, eliminating from the redistributed joint time-frequency domain elements that correspond to noise and interference, and recovering a signal devoid of noise and interference from the modified redistributed joint time-frequency domain.

The invention discloses a rotor rub impact fault detection method based on nonlinear compression conversion and a rotor rub impact fault detection system based on nonlinear compression conversion, and the method and the system are used for detecting the rub impact fault of a rotor system in a rotating machine. The method comprises the following steps of performing nonlinear compression conversion on a vibration signal of the rotor system so as to obtain time-frequency representation of the vibration signal; calculating the instantaneous frequency of the vibration signal in combination with the time-frequency representation; further calculating the frequency spectrum of a vibration part of the instantaneous frequency; if fr is the rotating frequency of the rotor system, the maximal spectrum peak of the frequency spectrum is within a range of [0.99fr,1.01fr] and the amplitude value is greater than 2% of the rotating frequency, judging that the rotor system has a rub impact fault; otherwise, judging that the rotor system does not has a rub impact fault. The system is implemented on the basis of the method and is capable of facilitating application of the method. By virtue of the method and the system, whether the rub impact fault exists or not can be accurately judged, and the result is rapid and reliable; the method and the system are simple and easy and are applicable to online health monitoring of a rotor.

Disclosed is method of generating an audible signal in a hearing aid by estimating a weighting function of received audio signals, the hearing aid is adapted to be worn by a user; the method comprises the steps of:estimating a directional signal by estimating a weighted sum of two or more microphone signals from two or more microphones, where a first microphone of the two or more microphones is a front microphone, and where a second microphone of the two or more microphones is a rear microphone;estimating a direction-dependent time-frequency gain, and synthesizing an output signal;wherein estimating the direction-dependent time-frequency gain comprises:obtaining at least two directional signals each containing a time-frequency representation of a target signal and a noise signal; and where a first of the directional signals is defined as a front aiming signal, and where a second of the directional signals is defined as a rear aiming signal;using the time-frequency representation of the target signal and the noise signal to estimate a time-frequency mask; andusing the estimated time-frequency mask to estimate the direction-dependent time-frequency gain.

In accordance with an embodiment, a time-frequency post-processing method of improving perceptual quality of a decoded audio signal, the method includes determining a time-frequency representation (such as filter bank analysis and synthesis) of an audio signal, estimating a time-frequency energy distribution of an audio signal from a time-frequency filter bank, computing a modification gain for each time-frequency representation point to have a modified time-frequency representation, and outputting audio signal from a modified time-frequency representation.

An audio decoder for providing a decoded audio information on the basis of an encoded audio information includes a window-based signal transformer configured to map a time-frequency representation, which is described by the encoded audio information, to a time-domain representation. The window-based signal transformer is configured to select a window, out of a plurality of windows including windows of different transition slopes and windows of different transform length, on the basis of a window information. The audio decoder includes a window selector configured to evaluate a variable-codeword-length window information in order to select a window for a processing of a given portion of the time-frequency representation associated with a given frame of the audio information.

This invention pertains to the extraction of the slowness dispersion characteristics of acoustic waves received by an array of two or more sensors by the application of a continuous wavelet transform on the received array waveforms (data). This produces a time-frequency map of the data for each sensor that facilitates the separation of the propagating components thereon. Two different methods are described to achieve the dispersion extraction by exploiting the time frequency localization of the propagating mode and the continuity of the dispersion curve as a function of frequency. The first method uses some features on the modulus map such as the peak to determine the time locus of the energy of each mode as a function of frequency. The second method uses a new modified Radon transform applied to the coefficients of the time frequency representation of the waveform traces received by the aforementioned sensors. Both methods are appropriate for automated extraction of the dispersion estimates from the data without the need for expert user input or supervision.

A listening device includes an ear-part for being worn in or at an ear of a user, a microphone system including at least two microphones each converting an input sound to an electrical microphone signal, and a TF-conversion unit for providing a time-frequency representation of the at least two microphone signals. Each signal representation includes complex or real values of the signal in a particular time-frequency unit. The listening device also includes a DIR-unit with a directionality system providing a weighted sum of the at least two electrical microphone signals thereby providing at least two directional microphone signals having maximum sensitivity in spatially different directions and a combined microphone signal. Each time-frequency unit of the combined signal is attributable to a particular direction. A frequency shaping-unit modifies one or more selected time-frequency units to indicate directional cues of input sounds providing an improved directional output signal.