Abstract: Techniques for improving adaptive interference cancellation (AIC) using cascaded AIC algorithms are described. To improve an accuracy of detecting speech, a device may perform a first stage of AIC to generate isolated audio data and may generate speech mask data indicating time windows when speech is detected in the isolated audio data. Based on the speech mask data, the device may perform second AIC to generate output audio data, with adaptation of the adaptive filter enabled when the speech is not detected and disabled when the speech is detected. Thus, the first AIC improves the accuracy with which the device detects that speech is present and the second AIC reduces distortion in the output audio data by not updating filter coefficient values when the speech is present. The first AIC may use playback audio data, microphone audio data or beamformed audio data as reference signals.

Abstract: Embodiments provide an apparatus for providing a fingerprint of an input signal, wherein the apparatus is configured to determine intensity values for a plurality of time-frequency regions of the input signal, wherein the apparatus is configured to compare the intensity values associated with different time-frequency regions of the plurality of time-frequency regions, to obtain individual values of the fingerprint based on the comparison of intensity values associated with two time-frequency regions.

Abstract: Embodiments described herein provide for automatically classifying the types of devices that place calls to a call center. A call center system can detect whether an incoming call originated from voice assistant device using trained classification models received from a call analysis service. Embodiments described herein provide for methods and systems in which a computer executes machine learning algorithms that programmatically train (or otherwise generate) global or tailored classification models based on the various types of features of an audio signal and call data. A classification model is deployed to one or more call centers, where the model is used by call center computers executing classification processes for determining whether incoming telephone calls originated from a voice assistant device, such as Amazon Alexa® and Google Home®, or another type of device (e.g., cellular/mobile phone, landline phone, VoIP).

Abstract: An audio event detection method including performing a framing processing on an audio to obtain audio data for each time period in the audio and extracting a specified feature vector from the audio data of each time period; inputting the specified feature vector of the audio data to a Recurrent Neural Network/Bidirectional Recurrent Neural Network (RNN/BI-RNN) model, to obtain a posterior probability of each pre-set audio event in the audio data of each time period; obtaining, for each time period, a target audio event of the audio data according to the posterior probability of each audio event in the audio data and a pre-set audio decoding algorithm; and extracting an optimal audio data sequence of the target audio event from the audio data of each time period.

Abstract: The present disclosure relates to a method and device for selecting from a plurality of beams. The method includes: obtaining a plurality of beam data, and performing frequency sampling on each of the plurality of beam data; obtaining a plurality of beam frequency correlation coefficients based on frequency sampling data of each of the plurality of beam data, in which a beam frequency correlation coefficient is configured to indicate a similarity between one in the plurality of beam data and another one in the plurality of beam data; obtaining a beam frequency correlation coefficient sum corresponding to each of the plurality of beam data based on the plurality of beam frequency correlation coefficients; and selecting beam data having the beam frequency correlation coefficient sum satisfying a preset correlation coefficient requirement in the plurality of beam data as target beam data.

Abstract: A system that acquires first audio data including a voice command captured by a microphone; identifies second audio data included in broadcast content corresponding to a timing at which the first audio data is captured by the microphone; extracts the second audio data from the first audio data to generate third audio data; converts the third audio data to text data corresponding to the voice command; and outputs the text data.

Abstract: A device capable of autonomous motion may move in response to a user speaking an utterance, such as a command. Before moving, the device processes audio data received from a microphone array to identify different audio signals arriving at the device from different directions. Based on properties of the audio signals, the device determines which of the audio signals are merely reflections of other audio.

Abstract: In one aspect, a first playback device is configured to (i) receive a set of voice signals, (ii) process the set of voice signals using a first set of audio processing algorithms, (iii) identify, from the set of voice signals, at least two voice signals that are to be further processed, (iv) determine that the first playback device does not have a threshold amount of computational power available, (v) receive an indication of an available amount of computational power of a second playback device, (vi) send the at least two voice signals to the second playback device, (vii) cause the second playback device to process the at least two voice signals using a second set of audio processing algorithms, (viii) receive, from the second playback device, the processed at least two voice signals, and (ix) combine the processed at least two voice signals into a combined voice signal.

Abstract: In one embodiment, an audio system can generate a parametrically formulated noise signal which can be adaptively reconfigured according to an input signal. According to an embodiment, an audio system can adaptively adjust a parametrically formulated noise signal according to environmental noise detected by the audio system. According to an embodiment, an audio system can present an adaptive level of activation energy in the presence of environmental noise such that a substantially constant and sufficient level of activation energy can be presented to an individual's auditory system such that additional sound energy corresponding to speech can become audible and intelligible to the individual.

Abstract: Techniques for non-linear acoustic echo cancellation are described herein. In an embodiment, a system comprises a loudspeaker, a microphone array, a spatial filtering logic with a spatial filter, an acoustic echo canceller (AEC) logic and an adder logic block. The spatial filtering logic is configured to generate a spatially-filtered signal by applying the spatial filter using a reference signal sent to the loudspeaker and a multi-channel microphone signal from the microphone array. The generated spatially-filtered signal carries both linear echo and non-linear echo that are included in the multi-channel microphone signal. The AEC logic is configured to apply a linear adaptive filter using the spatially-filtered signal to generate a cancellation signal that estimates both the linear echo and the non-linear echo of the multi-channel microphone signal. The adder logic block is configured to generate an output signal based on the cancellation signal.

Abstract: A voice processing device includes: a sound source localization unit configured to determine a direction of each sound source on the basis of voice signals of a plurality of channels; a sound source separation unit configured to separate signals for respective sound sources indicating components of respective sound sources from the voice signals of the plurality of channels; a speech section detection unit configured to detect a speech section in which the number of speakers is 1 from the signals for respective sound sources; and a speaker identification unit configured to identify a speaker on the basis of the signals for respective sound sources in the speech section.

Abstract: Systems and methods for audio signal processing including an input interface to receive a noisy audio signal including a mixture of target audio signal and noise. An encoder to map each time-frequency bin of the noisy audio signal to one or more phase-related value from one or more phase quantization codebook of phase-related values indicative of the phase of the target signal. Calculate, for each time-frequency bin of the noisy audio signal, a magnitude ratio value indicative of a ratio of a magnitude of the target audio signal to a magnitude of the noisy audio signal. A filter to cancel the noise from the noisy audio signal based on the phase-related values and the magnitude ratio values to produce an enhanced audio signal. An output interface to output the enhanced audio signal.

Abstract: A transfer function generation apparatus includes: a modeling part that models, using a function which uses an arrival direction of a sound source as a non-discrete argument, a plurality of acoustic transfer functions to a microphone from sound sources present in a plurality of directions and that stores the modeled function; and a transfer function generation part that generates a transfer function of an arbitrary direction by using the modeled and stored function.

Abstract: A method for generating and providing an audio signal, including receiving a first audio signal via an external microphone of a headphone or earphone, and receiving a second audio signal via a wireless interface. The first audio signal includes a portion reproduced via loudspeakers. The second audio signal corresponds to the portion reproduced via loudspeakers and is received before the corresponding portion of the first audio signal. A propagation time difference is determined between the first audio signal and the second audio signal. The second audio signal is modified by adaptive filtering and temporal shifting such that the propagation time difference between the first and second modified audio signal is substantially compensated. The adaptive filtering models an acoustic transmission of the first audio signal and a modified second audio signal is obtained. The modified second audio signal is inverted, then it is provided via the headphone or earphone.

Abstract: A speech intelligibility enhancing system for enhancing speech to be outputted in a noisy environment, the system comprising: a speech input for receiving speech to be enhanced; a noise input for receiving real-time information concerning the noisy environment; an enhanced speech output to output said enhanced speech; and a processor configured to convert speech received from said speech input to enhanced speech to be output by said enhanced speech output, the processor being configured to: apply a spectral shaping filter to the speech received via said speech input; apply dynamic range compression to the output of said spectral shaping filter; and measure the signal to noise ratio at the noise input, wherein the spectral shaping filter comprises a control parameter and the dynamic range compression comprises a control parameter and wherein at least one of the control parameters for the dynamic range compression or the spectral shaping is updated in real time according to the measured signal to noise ratio.

Abstract: A system and method are presented for forming the excitation signal for a glottal pulse model based parametric speech synthesis system. The excitation signal may be formed by using a plurality of sub-band templates instead of a single one. The plurality of sub-band templates may be combined to form the excitation signal wherein the proportion in which the templates are added is dynamically based on determined energy coefficients. These coefficients vary from frame to frame and are learned, along with the spectral parameters, during feature training. The coefficients are appended to the feature vector, which comprises spectral parameters and is modeled using HMMs, and the excitation signal is determined.

Abstract: The invention relates to a background noise estimator and a method therein, for supporting sound activity detection in an audio signal segment. The method comprises reducing a current background noise estimate when the audio signal segment is determined to comprise music and the current background noise estimate exceeds a minimum value. This is to be performed when an energy level of an audio signal segment is more than a threshold higher than a long term minimum energy level, lt_min, which is determined over a plurality of preceding audio signal segments, or, when the energy level of the audio signal segment is less than a threshold higher than lt_min, but no pause is detected in the audio signal segment.

Abstract: An apparatus includes multiple microphones to generate audio signals based on sound of a far-field acoustic environment. The apparatus also includes a signal processing system to process the audio signals to generate at least one processed audio signal. The signal processing system is configured to update one or more processing parameters while operating in a first operational mode and is configured to use a static version of the one or more processing parameters while operating in the second operational mode. The apparatus further includes a keyword detection system to perform keyword detection based on the at least one processed audio signal to determine whether the sound includes an utterance corresponding to a keyword and, based on a result of the keyword detection, to send a control signal to the signal processing system to change an operational mode of the signal processing system.

Abstract: A frequency domain converter divides an input signal for each predetermined frame, and generates a first signal X(f, ?) for each first frequency division unit. A noise estimation signal generator generates a signal Y(f, ?) for each second frequency division unit wider than the first frequency division unit. A signal comparator calculates a representative value for each second frequency division unit based on the signal Y(f, ?) stored in a storage unit, and compares the representative value and the signal Y(f, ?) with each other for each second frequency division unit. A mask generator generates a mask M(f, ?), which determines a degree of suppression or emphasis for each first frequency division unit, based on a peak range of the signal X(f, ?), and a comparison result by the signal comparator. The mask application unit multiplies the signal X(f, ?) by the mask M(f, ?).

Abstract: According to one embodiment, a signal processing apparatus includes a processer. The processor separates a plurality of signals, which are received at different positions and come from different directions, by a separation filter. The processor estimates incoming directions of a plurality of separate signals respectively, and associates the plurality of separate signals with transmission sources of the plurality of signals. The processor associates either one of a first attribute and a second attribute with the separate signals which are associated with the transmission sources of the signals based on results of the estimation of the incoming directions in a first period, and add either one of first label information and second label information.

Abstract: A voice communication system is equipped with a voice encoder which classifies respective bits of a voice information bit string in accordance with the degree of importance which is the magnitude of auditory influence when there is an error therein, classifies a group of bits which are high in degree of importance into a core layer and classifies a group of bits which are not high into an extension layer and a voice decoder which decodes a voice by using the bit strings in both of the core layer and the extension layer on the basis of frequency that the error is detected by error detection processing and when the frequency is low and decodes the voice using all bits or only some bets in the core layer when the frequency is high.

Abstract: A set of signal measures is sent, wherein each signal measure in the set of signal measures corresponds to a respective audio signal received by a playback device in a media playback system and is processed based on a first set of audio processing algorithms. A plurality of signal measures is identified in the set of signal measures. Audio signals corresponding to the identified plurality of signal measures are processed by one or more devices in the media playback system to improve a signal measure of each of the audio signals. The audio signals are processed based on a second set of audio processing algorithms. The processed audio signals are combined into a combined audio signal.

Abstract: A system and method are presented for forming the excitation signal for a glottal pulse model based parametric speech synthesis system. The excitation signal may be formed by using a plurality of sub-band templates instead of a single one. The plurality of sub-band templates may be combined to form the excitation signal wherein the proportion in which the templates are added is dynamically based on determined energy coefficients. These coefficients vary from frame to frame and are learned, along with the spectral parameters, during feature training. The coefficients are appended to the feature vector, which comprises spectral parameters and is modeled using HMMs, and the excitation signal is determined.

Abstract: A set of core points is aggregated from a set of points extracted from a large document. A point and a core point each is a topic covered in the document. For the core point in the set of core points, a network of associations is constructed, where an association in the network includes an entity that has a relationship with the core point by virtue of having contributed data in the document that relates to the core point. From the contributed data, a sentiment value of the contributed data is computed, the sentiment value being indicative of a sentiment of the entity towards the core point. From a set of sentiment values corresponding to the associations in the network of associations, an overall sentiment value is computed for the core point. The overall sentiment values for each core point in the document is reported.

Abstract: A system that includes a microphone array comprising a plurality of microphones positioned at different locations, where the microphones output microphone signals. A beamformer is applied to the microphone output signals and is configured to control a gain that is applied to the microphone output signals. The gain is frequency dependent and is related to a mismatch in sensitivity between two or more of the microphones.

Abstract: A controller for controlling operation of a switching regulator including a modulator, a discontinuous conduction mode (DCM) controller, an audible DCM (ADCM) controller, and a sub-sonic discontinuous conduction mode (SBDCM) controller. The modulator generally operates in a continuous conduction mode. The DCM controller modifies operation to DCM during low loads. The ADCM controller detects when the switching frequency is less than a super-sonic frequency threshold and modifies operation to maintain the switching frequency at a super-sonic frequency level. The SBDCM controller detects a sub-sonic operating condition during ADCM operation and responsively inhibits operation of the ADCM mode controller to allow a SBDCM mode within a sub-sonic switching frequency range. The SBDCM operating mode allows for efficient connected standby operation. The SBDCM controller allows operation to return to other modes when the switching frequency increases above the subsonic level.

Abstract: A system extracting features from a time-varying signal comprising a computer processor and a computer readable medium having computer executable instructions for providing: a bank of bandpass filters; a module approximating the output of those filters with nonlinear components; a module representing a decorrelated projection of the output of the filters with nonlinear components; and a module representing the temporal derivative of the decorrelated information with nonlinear components.

Abstract: The present invention discloses a headset, a terminal, and a method for processing an audio signal based on the headset and the terminal. The method for processing an audio signal includes: acquiring, by a first microphone of a headset in a call process, a first audio signal including a call voice and background noise, and acquiring, by a second microphone of the headset in the call process, a second audio signal including the background noise; and transmitting, by using an external plug of the headset, the first audio signal and the second audio signal to a noise reduction chip built in a terminal, so that the noise reduction chip performs noise reduction processing on the first audio signal according to the second audio signal.

Abstract: A processing apparatus estimates a noise amplitude spectrum of noise included in a sound signal. The processing apparatus includes an amplitude spectrum calculation part configured to calculate an amplitude spectrum of the sound signal for each one of frames obtained from dividing the sound signal into units of time; and a noise amplitude spectrum estimation part configured to estimate the noise amplitude spectrum of the noise detected from the frame. The noise amplitude spectrum estimation part includes a first estimation part configured to estimate the noise amplitude spectrum based on a difference between the amplitude spectrum calculated by the amplitude spectrum calculation part and the amplitude spectrum of the frame occurring before the noise is detected, and a second estimation part configured to estimate the noise amplitude spectrum based on an attenuation function obtained from noise amplitude spectra of the frames occurring after the noise is detected.

Abstract: Systems and methods are provided for audio encoding. For example, first quantization encoding is performed on audio data associated with a current frame of an audio data stream to obtain first quantization-encoded data; second quantization encoding is performed on the audio data to obtain second quantization-encoded data; the first quantization-encoded data is coupled to the current frame of the audio data stream; and the second quantization-encoded data is coupled to a next frame of the audio data stream.

Abstract: A communication method is provided. The communication method includes generating a pseudo background sound signal based on a gradient pulse control signal, performing a computation of subtracting the pseudo background sound signal from an acoustic signal having a sound signal and a background sound signal including a gradient coil drive sound signal, the acoustic signal obtained by an input device configured to receive the voice of a subject, and outputting sound based on a result of the computation, wherein a parameter of generating the pseudo background sound signal is controlled to reduce the difference resulting from the subtraction.

Abstract: Differing first and second audio signal sample rates from first and second audio signals are matched to each other. If signal sample rates are different, a frame of samples of the first audio signal is duplicated. The duplicate copies are multiplied by a window function and its inverse to produce “windowed frames” first and last samples of which can be deleted or added to increase or decrease a frame rate.

Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for processing spoken query terms. In one aspect, a method includes performing speech recognition on an audio signal to select two or more textual, candidate transcriptions that match a spoken query term, and to establish a speech recognition confidence value for each candidate transcription, obtaining a search history for a user who spoke the spoken query term, where the search history references one or more past search queries that have been submitted by the user, generating one or more n-grams from each candidate transcription, where each n-gram is a subsequence of n phonemes, syllables, letters, characters, words or terms from a respective candidate transcription, and determining, for each n-gram, a frequency with which the n-gram occurs in the past search queries, and a weighting value that is based on the respective frequency.

Abstract: The present disclosure is directed towards a process for estimating the signal to noise ratio of a speech signal. The process may include receiving, at a computing device, a speech signal having a bitstream and a signal-to-noise ratio (“SNR”) associated therewith. The process may further include estimating the SNR directly from the bitstream or using a partial decoder that is configured to extract one or more parameters, the parameters including at least one of a fixed codebook gain, an adaptive codebook gain, a pitch lag, and a line spectral frequency (“LSF”) coefficient.

Abstract: The invention provides an electronic method for assessing student skills on a stringed musical instrument, based on comparing the digital signature of an expert template sound to the digital signature of a student's attempt to replicate the template on the stringed instrument. In particular the method employs in sequential series an audio input device, an analog-to-digital converter, an audio correlation decoder, a digital-to-analog converter, and an audio output device; in certain embodiments they are all under the control of the same central control processor, which also has access to a first addressable data memory including unprocessed emulations from a student, a logic filter circuit element, a second addressable data memory including emulations from the student that have been filtered to remove background noise and other extraneous elements, and an addressable data memory including expert templates for incremental components of stringed musical performance.

Abstract: Systems, apparatuses and methods for integrating adaptive noise cancellation (ANC) with communication features in an enclosure, such as an incubator, bed, and the like. Utilizing one or more error and reference microphones, a controller for a noise cancellation portion reduces noise within a quiet area of the enclosure. Voice communications are provided to allow external voice signals to be transmitted to the enclosure with minimized interference with noise processing. Vocal communications from within the enclosure may be processed to determine certain characteristics/features of the vocal communications. Using these characteristics, certain emotive and/or physiological states may be identified.

Abstract: Acoustic noise suppression is provided in multiple-microphone systems using Voice Activity Detectors (VAD). A host system receives acoustic signals via multiple microphones. The system also receives information on the vibration of human tissue associated with human voicing activity via the VAD. In response, the system generates a transfer function representative of the received acoustic signals upon determining that voicing information is absent from the received acoustic signals during at least one specified period of time. The system removes noise from the received acoustic signals using the transfer function, thereby producing a denoised acoustic data stream.

Abstract: A method of signal processing an input signal in a hearing aid to avoid entrainment, the hearing aid including a receiver and a microphone, the method comprising using an adaptive filter to measure an acoustic feedback path from the receiver to the microphone and adjusting an adaptation rate of the adaptive filter using an output from a filter having an autoregressive portion, the output derived at least in part from a ratio of a predictive estimate of the input signal to a difference of the predictive estimate and the input signal.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.

Abstract: A voice quality measurement device that measures voice quality of a decoded voice signal outputted from a voice decoder unit. The voice quality measurement device includes a packet buffer unit and a voice information monitoring unit. The packet buffer unit accumulates voice packets that arrive non-periodically as voice information, and outputs the voice information to the voice decoder unit periodically. The voice information monitoring unit monitors continuity of the voice information inputted to the voice decoder unit, and calculates an index of voice quality of the decoded voice signal that reflects acceptability of this continuity.

Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Abstract: Embodiments are described of a multi-block coding scheme for an audio signal to prevent partial collapse conditions from causing pre-echo compression artifacts. An audio codec includes a segmentation component partitioning the audio signal into a plurality of tiles, wherein each tile comprises data from a particular segment of time and a particular set of frequencies of the audio signal; a band energy component determining an energy value for each tile corresponding to a signal component in a respective tile; an encoder flag tracking component marking a tile as not collapsed or collapsed based on the energy value in that tile; and a decoder flag tracking component filling all tiles marked as collapsed with pseudorandom noise at an estimated energy level.

Abstract: A howling canceller which suppresses occurrence of howling even when an open loop gain exceeds “1” in the whole reproduction band. In the howling canceller, an adaptive filter (107) operates a digital received voice signal with a tap coefficient to generate a pseudo echo; a subtractor (108) subtracts the pseudo echo from a digital transmitted voice signal to generate a residual signal; and an amplitude limiting circuit (110) limits the absolute value of the amplitude of the digital received voice signal to be equal to or smaller than a predetermined threshold which ensures that all of a D/A converter (101), a power amplifier (102), a speaker (103), a microphone (104), a microphone amplifier (105), and an A/D converter (106) operate in a linear operation area, and outputs the amplitude-limited digital received voice signal to the D/A converter (101) and the adaptive filter (107).

Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.

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 portions 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: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal.

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: The present invention provides a noise reduction control method using a microphone array and a noise reduction control device using a microphone array wherein the method comprises the steps of: S1: collecting, by the microphone array, acoustic signals; S2: estimating incidence angles of all acoustic signals of the microphone array; S3: conducting a statistics on signal components according to incidence angles; S4: determining a parameter ? from a ratio of noise components according to the statistical result and using the parameter ? as a control parameter for controlling an adaptive filter. With the present invention, space position information of the sound is obtained directly with the microphone array to control update of the adaptive filter more accurately, so as to eliminate noise, enhance SNR and protect speech quality well at the same time.