Abstract: A method for performing packet loss or Frame Erasure Concealment (FEC) for a speech coder receives encoded frames of compressed speech information transmitted from an encoder. The method determines whether an encoded frame has been lost, corrupted in transmission, or erased, synthesizes properly received frames, and decides on an overlap-add window to use in combining a portion of the synthesized speech signal with a subsequent speech signal resulting from a received and decoded packet, where the size of the overlap-add window is based on the unavailability of packets. If it is determined that an encoded frame has been lost, corrupted in transmission, or erased, the method performed an overlap-add operation on the portion of the synthesized speech signal and the subsequent speech signal, using the decided-on overlap-add window.

Abstract: A voice recognition system comprises: a voice input unit that receives an input signal from a voice input element and output it; a voice detection unit that detects an utterance segment in the input signal; a voice recognition unit that performs voice recognition for the utterance segment; and a control unit that outputs a control signal to at least one of the voice input unit and the voice detection unit and suppresses a detection frequency if the detection frequency satisfies a predetermined condition.

Abstract: The invention proposes the synthesis of a signal consisting of consecutive blocks. It proposes more particularly, on receipt of such a signal, to replace, by synthesis, lost or erroneous blocks of this signal. To this end, it proposes an attenuation of the overvoicing during the generation of a signal synthesis. More particularly, a voiced excitation is generated on the basis of the pitch period (T) estimated or transmitted at the previous block, by optionally applying a correction of plus or minus a sample of the duration of this period (counted in terms of number of samples), by constituting groups (A?,B?,C?,D?) of at least two samples and inverting positions of samples in the groups, randomly (B?,C?) or in a forced manner. An over-harmonicity in the excitation generated is thus broken and the effect of overvoicing in the synthesis of the generated signal is thereby attenuated.

Abstract: The present invention relates to signal modification before pitch period repetition for the synthesis of blocks lost on decoding digital audio signals. The effects of repetition of transitories, such as the plosives of a speech signal, are avoided by comparing the samples of a pitch period with those of the previous pitch period. The signal is modified preferentially by taking the minimum between a current sample (e(3)) of the last pitch period (Tj) and at least one sample (e(2?T0) of approximately the same position in the previous pitch period (Tj?1).

Abstract: A method to audio matrix encode/decode, which encode and decode audio signals of two or more channels into an audio signal of one or more channel while preserving the direction of a sound image includes extracting pieces of sound image information from audio signals of multi channels, encoding and allocating the extracted sound image information to an inaudible frequency domain except an audible frequency domain, and adding the sound image information allocated to the inaudible frequency domain and matrix-encoded stereo signals of the audible frequency domain.

Abstract: A system and method for enhancing a tonal sound signal decoded by a decoder of a speech-specific codec in response to a received coded bit stream, in which a spectral analyser is responsive to the decoded tonal sound signal to produce spectral parameters representative of the decoded tonal sound signal. A quantization noise in low-energy spectral regions of the decoded tonal sound signal is reduced in response to the spectral parameters produced by the spectral analyser. The spectral analyser divides a spectrum resulting from spectral analysis into a set of critical frequency bands each comprising a number of frequency bins, and the reducer of quantization noise comprises a noise attenuator that scales the spectrum of the decoded tonal sound signal per critical frequency band, per frequency bin, or per both critical frequency band and frequency bin.

Abstract: A method is provided for concealing a transmission error in a digital signal chopped into a plurality of successive frames associated with different time intervals in which, on reception, the signal may comprise erased frames and valid frames, the valid frames comprising information relating to the concealment of frame loss. The method is implemented during a hierarchical decoding using a core decoding and a transform-based decoding using windows introducing a time delay of less than a frame with respect to the core decoding. The method includes concealing a first set of missing samples for the erased frame, implemented in a first time interval; a step of concealing a second set of missing samples utilizing information of said valid frame and implemented in a second time interval; and a step of transition between the first and the second set of missing samples to obtain at least part of the missing frame.

Abstract: A system is described that performs frame erasure concealment to generate frames of an output speech signal corresponding to a series of erased frames of encoded bit-stream in a manner that conceals the quality-degrading effects of such erased frames. In one embodiment, responsive to the detection of a first erased frame in the series, a number of steps are performed. These steps include deriving long-term and short synthesis filters based on previously-generated portions of the output speech signal, calculating a ringing signal segment based on the long-term and short-term synthesis filters, and generating a frame of the output speech signal corresponding to the first erased frame by overlap adding the ringing signal segment to an extrapolated waveform. Deriving the long-term filter includes estimating a pitch period based on a previously-generated portion of the output speech signal by finding a lag that minimizes a sum of magnitude difference function.

Abstract: A method and apparatus for concealing frame loss and an apparatus for transmitting and receiving a speech signal that are capable of reducing speech quality degradation caused by packet loss are provided. In the method, when loss of a current received frame occurs, a random excitation signal having the highest correlation with a periodic excitation signal (i.e., a pitch excitation signal) decoded from a previous frame received without loss is used as a noise excitation signal to recover an excitation signal of a current lost frame. Furthermore, a third, new attenuation constant (AS) is obtained by summing a first attenuation constant (NS) obtained based on the number of continuously lost frames and a second attenuation constant (PS) predicted in consideration of change in amplitude of previously received frames to adjust the amplitude of the recovered excitation signal for the current lost frame.

Abstract: The present invention describes a speech enhancement method using microphone arrays and a new iterative technique for enhancing noisy speech signals under low signal-to-noise-ratio (SNR) environments. A first embodiment involves the processing of the observed noisy speech both in the spatial- and the temporal-domains to enhance the desired signal component speech and an iterative technique to compute the generalized eigenvectors of the multichannel data derived from the microphone array. The entire processing is done on the spatio-temporal correlation coefficient sequence of the observed data in order to avoid large matrix-vector multiplications. A further embodiment relates to a speech enhancement system that is composed of two stages. In the first stage, the noise component of the observed signal is whitened, and in the second stage a spatio-temporal power method is used to extract the most dominant speech component.

Abstract: An apparatus and method for processing an audio signal including extracting noise filling flag information indicating whether noise filling is used to a plurality of frames; extracting coding scheme information indicating whether a current frame included in the plurality of frames is coded in either a frequency domain or a time domain; when the noise filling flag information indicates that the noise filling is used for the plurality of frames and the coding scheme information indicates that the current frame is coded in the frequency domain, extracting noise level information for the current frame; when a noise level value corresponding to the noise level information meets a predetermined level, extracting noise offset information for the current frame; and, when the noise offset information is extracted, performs the noise-filling for the current frame based on the noise level value and the noise offset information.

Abstract: A method and corresponding apparatus for coded-domain acoustic echo control is presented. An echo control problem is considered as that of perceptually matching an echo signal to a reference signal. A perceptual similarity function that is based on the coded spectral parameters produced by the speech codec is defined. Since codecs introduce a significant degree of non-linearity into the echo signal, the similarity function is designed to be robust against such effects. The similarity function is incorporated into a coded-domain echo control system that also includes spectrally-matched noise injection for replacing echo frames with comfort noise. Using actual echoes recorded over a commercial mobile network, it is shown herein that the similarity function is robust against both codec non-linearities and additive noise. Experimental results further show that the echo-control is effective at suppressing echoes compared to a Normalized Least Mean Squared (NLMS)-based echo cancellation system.

Abstract: An audio signal processing apparatus, includes an environmental ambient noise level detection unit for detecting an environmental ambient noise level contained in an audio signal inputted through sound collection means for collecting a transmission sound at the time of a voice call, a signal level adjustment unit which has a level adjustment function to adjust an output signal level with respect to an input signal level, and an input/output characteristic change function to change an input/output characteristic when adjusting a level in the level adjustment function by means of a control signal, and in which a received sound signal in the case of the telephone call voice is arranged to be an input signal, and a control signal generation unit for generating the control signal for changing the input/output characteristic of the signal level adjustment unit from the environmental ambient noise level detected by the environmental ambient noise level detection unit.

Abstract: A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal.

Abstract: The present invention discloses a speech enhancing method, a speech enhancing device and a denoising communication headphone.

Abstract: Wideband speech signals must be converted to narrowband speech signals if the transmission medium or the destination terminal is constructed with narrowband constraints. A typical wideband-to-narrowband conversion method is the elimination of frequencies above 3400 Hz using a low pass filter and a down sampler. However, this method produces a muffled speech sound since the resulting narrowband signal has a flat frequency response. Methods and apparatus are presented herein to enhance the acoustic quality of a wideband-to-narrowband converted signal. A bandwidth switching filter is used to emphasize a mid-range frequency portion of the wideband signal so that the resulting narrowband signal has a non-flat frequency spectrum.

Abstract: A device for lost frame concealment comprises: a lost frame detector for detecting whether a voice frame is lost, a decoding module for decoding the current voice frame, a low band delay module for delaying the low band signal, a low band signal recovering module for recovering the lost low band signal, a high band lost frame concealment module for processing the lost frame concealment for the high band signal, and a QMF synthesis filter for synthetically filtering the low band signal and the high band signal. The invention makes full use of the delay of the coding/decoding device itself, enhances the effect of lost frame concealment for the low band signal and the high band signal, and introduces no nearby delay during the process of lost frame concealment.

Abstract: A packet loss concealment method and system is described that attempts to reduce or eliminate destructive interference that can occur when an extrapolated waveform representing a lost segment of a speech or audio signal is merged with a good segment after a packet loss. This is achieved by guiding a waveform extrapolation that is performed to replace the bad segment using a waveform available in the first good segment or segments after the packet loss. In another aspect of the invention, a selection is made between a packet loss concealment method that performs the aforementioned guided waveform extrapolation and one that does not. The selection may be made responsive to determining whether the first good segment or segments after the packet loss are available and also to whether a segment preceding the lost segment and the first good segment following the lost segment are deemed voiced.

Abstract: A first matrix (W(k)) indicating frequency characteristics of a separation filter is calculated from input signals of a plurality of channels. A second matrix (Ws(k)) is calculated by using the restriction coefficients (Ci(k)) for restricting the separation filter and the first matrix, and separation filter coefficients (wsij(s)) are calculated by using the second matrix. With use of the separation filter coefficients, separation signals (ysi(t)) are then calculated from the input signals. A third matrix (Ws?1(k)) is then calculated by transforming the second matrix into an inverse matrix at each frequency, and reproduction filter coefficients (a?I1(s), a?I2(s)) are calculated by using the third matrix. With use of the reproduction filter coefficients, the synthesized signal of each channel is calculated by using the separation signals.

Abstract: An echo suppressing system includes: a sound output device for outputting sound based on a sound signal, including a passing section for allowing passage of a component of a different frequency band, and a plurality of sound output sections, each of which outputs sound based on each of the plurality of sound signals passed through the passing section; a summer for summing the plurality of sound signals to generate a reference sound signal; a sound input device for converting input sound into a sound signal; and an echo suppressor for suppressing echo based on the sound output by the sound output device, including an input section to which a sound signal is input from the sound input device as an observation sound signal, and a correction section for correcting the observation sound signal so as to suppress echo included in the observation sound signal.

Abstract: A system and method is described for compensating for the effects of a corrupted Continuously Variable Delta Slope Modulation (CVSD) decoder memory state on a decoded audio signal. In accordance with the system and method, a first estimated step size associated with a first frame of the decoded audio signal is calculated and a second estimated step size associated with a replacement frame generated to conceal bit errors in the first frame of the decoded audio signal is calculated. At least a second frame of the decoded audio signal is then modified based on the first estimated step size and the second estimated step size.

Abstract: A method for speech switching, including: extracting mute flags from encoded speech data transmitted by each of the terminals respectively in order to determine one or more non-mute terminals; decoding the encoded speech data of each non-mute terminal respectively; calculating speech energy of each non-mute terminal according to the decoded speech data of each non-mute terminal; comparing the speech energy of the non-mute terminals and selecting one or more terminals with relatively large speech energy; performing linear superposition with different combination methods to the decoded speech data of the selected terminals, and encoding the decoded speech data which is obtained from the linear superposition of the different combination methods, and transmitting the data to the corresponding terminals respectively. An apparatus for speech switching is also disclosed. With this invention, resource consumption during procedure of speech switching may be reduced, the effect of speech communication may be improved.

Abstract: A system and method for processing a narrowband speech signal comprising speech samples in a first range of frequencies. the method comprises: generating from the narrowband speech signal a highband speech signal in a second range of frequencies above the first range of frequencies; determining a pitch of the highband speech signal; using the pitch to generate a pitch-dependent tonality measure from samples of the highband speech signal; and filtering the speech samples using a gain factor derived from the tonality measure and selected to reduce the amplitude of harmonics in the highband speech signal.

Abstract: A noise reducing apparatus includes: a voice signal inputting unit inputting an input voice signal; a noise occurrence period detecting unit detecting a noise occurrence period; a noise removing unit removing a noise for the noise occurrence period; a generation source signal acquiring unit acquiring a generation source signal with a time duration corresponding to a time duration corresponding to the noise occurrence period; a pitch calculating unit calculating a pitch of an input voice signal interval; an interval signal setting unit setting interval signals divided in each unit period interval; an interpolation signal generating unit generating an interpolation signal with the time duration corresponding to the noise occurrence period and alternately arranging the interval signal in a forward time direction and the interval signal in a backward time direction; and a combining unit combining the interpolation signal and the input voice signal, from which the noise is removed.

Abstract: A voice activity detection process is robust to a low and high signal-to-noise ratio speech and signal loss. A process divides an aural signal into one or more bands. Signal magnitudes of frequency components and the respective noise components are estimated. A noise adaptation rate modifies estimates of noise components based on differences between the signal to the estimated noise and signal variability.

Abstract: A sound data decoding apparatus based on a waveform coding method includes a loss detector, sound data decoder, sound data analyzer, parameter modifying section and sound synthesizing section. The loss detector detects whether a loss exists in a sound data. The sound data decoder decodes the sound data to generate a first decoded sound signal. The sound data analyzer extracts a first parameter from the first decoded sound signal. The parameter modifying section modifies the first parameter based on a result of the detection of loss. The sound synthesizing section generates a first synthesized sound signal by using the modified first parameter. Thus, a deterioration of sound quality is prevented in the error compensation of sound data.

Abstract: Packet loss concealment (PLC) systems and methods are described that use time-warping to merge a concealment signal generated to replace one or more bad frames of an audio signal with a received signal representing one or more subsequent good frames of the audio signal in a manner that avoids signal discontinuity and audible artifacts resulting therefrom. Prediction-based PLC systems and methods are also described that use time-warping to conceal the loss of one or more frames containing a transition region in a manner that will not result in an audible artifact.

Abstract: An audio signal quality enhancement apparatus and method. The apparatus includes a pitch calculating unit to extract a pitch period of an audio signal, a frequency domain transforming unit to transform the audio signal to a frequency domain, a frequency band dividing unit to classify the transformed audio signal into audio signals for each of the plurality of frequency bands based on the extracted pitch period, and a pitch enhancement unit to determine a gain based on a volume of the transformed audio signal, and to generate an output signal by multiplying each of the classified audio signals with respect to each of the plurality of frequency bands by the gain, thereby enhancing quality of the audio signal.

Abstract: A post filter and a decoder enabling improvement of the sound quality of a decoded signal even when the sound quality of the decoded signal is different from the bands are disclosed. A frequency converting section determines a decoded spectrum. A power spectrum computing section computes the power spectrum from the decoded spectrum. A correction band determining section determines the band in which the power spectrum is corrected according to layer information. A power spectrum correcting section corrects the power spectrum in the corrected band in such a way that the variation along the frequency axis is suppressed. An inverse converting section subjects the corrected power spectrum to inverse conversion to determine an autocorrelation function. An LPC analyzing section determines an LPC coefficient of the determined autocorrelation function.

Abstract: A post-filtering apparatus and method for speech enhancement in a modified discrete cosine transform (MDCT) domain are disclosed. In the apparatus and method, previous and current MDCT coefficients are used for obtaining a speech spectrum coefficient similar to a real speech spectrum, and a convex function is used for transforming the speech spectrum coefficient and obtaining a post-filter coefficient so that difference can increase in the case where the speech spectrum coefficient is small but decrease in the case where the coefficient is large. Then, the post-filter coefficient is applied to the MDCT coefficient. With this configuration, both the current and previous MDCT values are used, so that it is possible to obtain a spectrum coefficient similar to the real speech spectrum and to obtain a more accurate filter coefficient. Further, the coefficient is adaptively transformed through the convex function, thereby enhancing speech quality.

Abstract: A bit error concealment (BEC) system and method is described herein that detects and conceals the presence of click-like artifacts in an audio signal caused by bit errors introduced during transmission of the audio signal within an audio communications system. A particular embodiment of the present invention utilizes a low-complexity design that introduces no added delay and that is particularly well-suited for applications such as Bluetooth® wireless audio devices which have low cost and low power dissipation requirements.

Abstract: A noise cancellation apparatus includes a noise estimation module for receiving a noise-containing input speech, and estimating a noise therefrom to output the estimated noise; a first Wiener filter module for receiving the input speech, and applying a first Wiener filter thereto to output a first estimation of clean speech; a database for storing data of a Gaussian mixture model for modeling clean speech; and an MMSE estimation module for receiving the first estimation of clean speech and the data of the Gaussian mixture model to output a second estimation of clean speech. The apparatus further includes a final clean speech estimation module for receiving the second estimation of clean speech from the MMSE estimation module and the estimated noise from the noise estimation module, and obtaining a final Wiener filter gain therefrom to output a final estimation of clean speech by applying the final Wiener filter gain.

Abstract: A system improves the speech intelligibility and the speech quality of a speech segment. The system includes a dynamic controller that detects a background noise from an input by modeling a signal. A variable gain amplifier adjusts the variable gain of the amplifier in response to an output of dynamic controller. A shaping filter adjusts a speech signal by tilting portions of the speech signal of the dynamic controller.

Abstract: An audio signal is conveyed more efficiently by transmitting or recording a baseband of the signal with an estimated spectral envelope and a noise-blending parameter derived from a measure of the signal's noise-like quality. The signal is reconstructed by translating spectral components of the baseband signal to frequencies outside the baseband, adjusting phase of the regenerated components to maintain phase coherency, adjusting spectral shape according to the estimated spectral envelope, and adding noise according to the noise-blending parameter. Preferably, the transmitted or recorded signal also includes an estimated temporal envelope that is used to adjust the temporal shape of the reconstructed signal.

Abstract: In a noise reduction apparatus for controlling noise up to a predetermined upper limited frequency, a distance from a noise source to control point X is made larger than a distance obtained by subtracting a one-half wavelength from a distance, obtained by adding up a distance from the noise source to a noise detecting microphone, a distance corresponding to time as a sum of respective delay time of the noise detecting microphone, a noise controller, and a control speaker, and a distance from the control speaker to control point X, where one wavelength is a period corresponding to the upper limited frequency.

Abstract: Systems and methods are described for performing packet loss concealment using an extrapolation of an excitation waveform in a sub-band predictive speech coder, such as an ITU-T Recommendation G.722 wideband speech coder. The systems and methods are useful for concealing the quality-degrading effects of packet loss in a sub-band predictive coder and address some sub-band architectural issues when applying excitation extrapolation techniques to such sub-band predictive coders.

Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between segments of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.

Abstract: A double talk detector for controlling the echo path estimation in a telecommunication system by indicating when a received coded speech signal is dominated by a non-echo signal; i.e., that so-called double talk exists. This is determined by extracting LSPs from a coded speech frame of the received coded speech signal when the signal power exceeds a first threshold value, converting each of said extracted LSPs into LSFs, and calculating the distance between each two adjacent LSFs. For each distance that is smaller than a second threshold, a spectral peak is located between the two LSFs, and it is determined whether said spectral peak is an echo or not. When a predetermined number of non-echo spectral peaks are located in the received speech signal, double talk will be indicated, and the echo path estimation may be disabled.

Abstract: A basic idea of the invention is to ascertain information on the course of the bit rate switching during an active speech phase. According to the invention, during the speech phase, information on the percentage proportion of broadband active speech frames in comparison to narrowband active speech frames is compiled on the part of the decoder. A high percentage proportion of broadband active speech frames indicates that a broadband use is preferred on the part of the codec and therefore a need exists for synthesizing noise information in broadband form during a DTX phase.

Abstract: An enhancement system improves the estimate of noise from a received signal. The system includes a spectrum monitor that divides a portion of the signal at more than one frequency resolution. Adaptation logic derives a noise adaptation factor of the received signal. A plurality of devices tracks the characteristics of an estimated noise in the received signal and modifies multiple noise adaptation rates. Weighting logic applies the modified noise adaptation rates derived from the signal divided at a first frequency resolution to the signal divided at a second frequency resolution.

Abstract: An audio/music decoding device capable of improving quality of a decoded signal generated by conceal processing of a frame erase using a scalable encoding method. The audio/music decoding device includes a frame loss detector that determines whether encoded information is normally received and generates frame loss information indicating the result of the determination. According to the frame loss information, a first decoder performs decoding by using at least one of the following encoded information: the first encoded information on the frame immediately before, the first encoded information on the current frame, and the second encoded information on the current frame.

Abstract: A microphone-array-based speech recognition system using a blind source separation (BBS) and a target speech extraction method in the system are provided. The speech recognition system performs an independent component analysis (ICA) to separate mixed signals input through a plurality of microphone into sound-source signals, extracts one target speech spoken for speech recognition from the separated sound-source signals by using a Gaussian mixture model (GMM) or a hidden Markov Model (HMM), and automatically recognizes a desired speech from the extracted target speech. Accordingly, it is possible to obtain a high speech recognition rate even in a noise environment.

Abstract: A method of providing identifying information over a voice communications link can include receiving, from a call participant, a personal identification code over the voice communications link, determining identifying information for the call participant using the personal identification code, and encoding the identifying information of the call participant within a voice stream carried by the voice communications link. The voice stream and identifying information can be sent to a subscriber.

Abstract: A frame compensation method is provided. The method includes: obtaining a length of a lost frame and a length of a correct frame; determining that the length of the correct frame is integral power of 2 times of the length of the lost frame, and then obtaining a data sequence with the same length as the length of the lost frame according to the correct frame; and compensating the lost frame according to the data sequence to obtain a compensated data frame. A frame compensation system is also provided. Lost frames in various formats are compensated according to correct frames in various formats, so that the limitation of the related art that a lost frame in a single format can be merely compensated according to a correct frame in a single format is eliminated, and the effect of the compensated data frames is better than that of filling comfort noises.

Abstract: Presented herein are systems and methods for processing sound signals for use with electronic speech systems. Sound signals are temporally parsed into frames, and the speech system includes a speech codebook having entries corresponding to frame sequences. The system identifies speech sounds in an audio signal using the speech codebook.

Abstract: A technique is described herein for reducing audible artifacts in an audio output signal generated by decoding a received frame in a series of frames representing an encoded audio signal in a predictive coding system. In accordance with the technique, it is determined if the received frame is one of a predefined number of received frames that follow a lost frame in the series of the frames. Responsive to determining that the received frame is one of the predefined number of received frames, at least one parameter or signal associated with the decoding of the received frame is altered from a state associated with normal decoding. The received frame is then decoded in accordance with the at least one parameter or signal to generate a decoded audio signal. The audio output signal is then generated based on the decoded audio signal.

Abstract: An audio data transmitting/receiving apparatus for realizing a high-quality frame compensation in audio communications. In an audio data transmitting apparatus (10), a delay part (104) subjects multi-channel audio data to a delay process that delays the L-ch encoded data relative to the R-ch encoded data by a predetermined delay amount. A multiplexing part (106) multiplexes the audio data as subjected to the delay process. A transmitting part (108) transmits the audio data as multiplexed. In an audio data receiving apparatus (20), a separating part (114) separates, for each channel, the audio data received from the audio data transmitting apparatus (10). A decoding part (118) decodes, for each channel, the audio data as separated. If there has occurred a loss or error in the audio data as separated, then a frame compensating part (120) uses one of the L-ch and R-ch encoded data to compensate for the loss or error in the other encoded data.

Abstract: System and method for aggregation and monitoring of multimedia data stored in a decentralized way for triggering upcoming class action events, whereas source databases of the network node are accessed by a filter module, and for at least one rating parameter in connection with assigned search key words and/or the assigned source databases with respect to a time-based rating and an exposure-based frequency rating a scorecard is generated with found data sets, and a parameterization module, based on the scorecard for the respective rating parameter with respect to their exposure-based frequency a variable frequency value is generated at least partially dynamically, which variable frequency value corresponds to network class action frequency variations, with respect to time, and a tracing unit based on a generated assigned distribution of the variable frequency values a predefined exposure threshold is triggered.

Abstract: An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.

Abstract: Cross-channel interference is eliminated and multi-channel sources are separated by estimating a source absence probability for a current frame of a first channel output, and determining an interference elimination coefficient for matching a secondary signal of the first channel output with a primary signal of a second channel output by using the source absence probability, generating an interference signal by multiplying the second channel output by an over-subtraction factor and the interference elimination coefficient, wherein a partial differentiation is performed for a v-norm value of a spectral amplitude difference, between the first channel output and the second channel output multiplied by the interference elimination coefficient and a result of multiplication of the source absence probability, by using the interference elimination coefficient to determine an update amount of the interference elimination coefficient for a next frame.