Abstract: An apparatus includes a group of microphones and a surface compensator that is operatively coupled to switch logic and to a signal conditioner that provides a control channel to voice recognition logic. The surface compensator may detect surfaces in proximity to the apparatus as well as the surface's acoustic reflectivity or acoustic absorptivity and may accordingly configure the group of microphones including selecting appropriate signal conditioning and beamforming based on the surface acoustic reflectivity or acoustic absorptivity and the orientation of the apparatus. Voice recognition performance is thus improved when microphones are impeded or occluded by proximate surfaces. A group of sensors of the apparatus is used by the surface compensator to detect surfaces and surface type, and to determine apparatus orientation and motion.

Abstract: A method is presented for estimating and suppressing reverberation from a digital reverberant signal. A method for changing a first reverberation estimation according to another reverberation estimation is further provided. A method for controlling the reverberation suppression rate is also presented.

Abstract: In one form, an audio processor circuit includes a first digital signal processing circuit, a second digital signal processing circuit, and an interleaver. The first digital signal processing circuit has an input for receiving a far-end audio signal, and an output. The second digital signal processing circuit has an input for receiving a digital near-end audio signal, and an output. The interleaver has a first input coupled to the output of the first digital signal processing circuit, a second input coupled to the output of the second digital signal processing circuit, and an output for alternatively providing signals received from the first and second inputs to the output.

Abstract: A sound source separation apparatus includes a transfer function storage unit that stores a transfer function from a sound source, a sound change detection unit that generates change state information indicating a change of the sound source on the basis of an input signal input from a sound input unit, a parameter selection unit that calculates an initial separation matrix on the basis of the change state information generated by the sound change detection unit, and a sound source separation unit that separates the sound source from the input signal input from the sound input unit using the initial separation matrix calculated by the parameter selection unit.

Abstract: An acoustic space including a sound output path, first and second sound input paths is formed in a main body casing of an earphone microphone. Output sound from a speaker propagates in the sound output path. Sound input to a first microphone propagates in the first sound input path communicating with outside. Sound input to a second microphone propagates in the second sound input path. The sound output path branches into one path communicating with the outside of the main body casing and the other path communicating with the second sound input path. The earphone microphone amplifies a sound signal output from at least one of the first and second microphones so as to input sound from a sound source outside the main body casing, and suppresses input of the output sound.

Abstract: According to one embodiment, an information processing apparatus includes a first signal input unit configure to receive a first signal, a second signal input unit configure to receive a signal, a first control unit configure to acquire system resources, a second control unit configure to select, in accordance with information of the system resources acquired by the first control unit, a processing method for suppressing at least one of echo and noise of the second signal input from the second signal input unit containing the echo due to the first signal input from the first signal input unit, a third control unit configure to generate an output signal by suppressing at least one of the echo and the noise from the second signal by the processing method selected by the second control unit, and a signal output unit configure to output the output signal generated by the third control unit.

Abstract: A telecommunication system including a fall duplex speakerphone, comprising a first microphone to generate a coupled signal including uplink information and non-linear distortion sensed by the first microphone in a speaker phone mode, a second microphone to generate a reference signal including downlink information and the non-linear distortion sensed by the second microphone in the speaker phone mode, and an acoustic echo canceller (AEC) to receive the coupled signal from the first microphone, to receive the reference signal from the second microphone, and to cancel out the non-linear distortion included in the coupled signal based on the non-linear distortion included in the reference signal.

Abstract: The present invention discloses an echo cancellation method. The method includes: dividing an audio signal into a high-band audio signal and a low-band audio signal; performing adaptive filtering on the low-band audio signal, and performing synthesis filtering on a signal obtained after the low-band audio signal undergoes the adaptive filtering and on the high-band audio signal to generate a preliminary echo cancellation signal; performing envelope predication echo suppression on a high-band signal in the preliminary echo cancellation signal, and calculating and outputting a residual echo suppression coefficient; performing echo suppression on a low-band signal in the preliminary echo cancellation signal, and outputting a processing result; and multiplying the output result by the residual echo suppression coefficient, and outputting a signal of which echoes are canceled.

Abstract: An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

Abstract: A method and an audio processing system determine a system parameter, e.g. step size, in an adaptive algorithm, e.g. an adaptive feedback cancellation algorithm so as to provide an alternative scheme for feedback estimation in a multi-microphone audio processing system. A feedback part of the system's open loop transfer function is estimated and separated in a transient part and a steady state part, which can be used to control the adaptation rate of the adaptive feedback cancellation algorithm by adjusting the system parameter, e.g. step size parameter, of the algorithm when desired system properties, such as a steady state value or a convergence rate of the feedback, are given/desired. The method can be used for different adaptation algorithms such as LMS, NLMS, RLS, etc. in hearing aids, headsets, handsfree telephone systems, teleconferencing systems, public address systems, etc.

Abstract: An signal processing apparatus, system and software product for audio modification/substitution of a background noise generated during an event including, but not be limited to, substituting or partially substituting a noise signal from one or more microphones by a pre-recorded noise, and/or selecting one or more noise signals from a plurality of microphones for further processing in real-time or near real-time broadcasting.

Abstract: A system and method for mitigating microphone hiss may obtain a frequency spectrum characteristic for a microphone. A microphone that has limited dynamic range may create microphone hiss in an output signal. The microphone hiss may prevent a reproduction of a sound field, represented in an output signal of the microphone, from being perceived as a natural environment. The microphone frequency spectrum may be obtained using static measurements or calculated dynamically. A virtual noise floor may be calculated responsive to the microphone frequency spectrum and a desired noise floor. Gain coefficients may be calculated responsive to the output signal of the microphone. The gain coefficients may be calculated to mitigate undesirable signal content including background noise and echoes. The calculated gain coefficients may be modified responsive to the virtual noise floor. The modified gain coefficients may allow a reproduction of the sound field to be perceived as a natural environment.

Abstract: An adaptive filter unit outputs a send-mid signal obtained by eliminating echo from a send-in signal, and a power comparing unit calculates a power ratio between received signal power and send-mid signal power. When a receiver ST detecting unit detects a single talk state at a receiving side, an acoustic coupling amount estimating unit estimates and updates the estimated amount of acoustic coupling from the power ratio. A residual echo power estimating unit estimates estimated residual echo power from the received signal power and the estimated amount of acoustic coupling, and a signal-to-echo ratio estimating unit estimates a ratio between the send-mid signal power and the estimated residual echo power. An amplitude suppression coefficient determining unit determines the amplitude suppression coefficient corresponding to the ratio, and an amplitude suppression unit amplitude suppresses the send-mid signal.

Abstract: An audio conferencing system has a base station, a speaker and one or more mobile microphones, and it operates to receive audio information from both a far-end audio source and a near-end audio source and to process the audio information for transmission to a far-end audio conferencing system. At least one of the mobile microphones has a motion detection device and operates to send detected motion information to the base station. The base station operates to control the processing of audio information it receives from the microphone according to the detected microphone motion information.

Abstract: An audio enhancement system includes a display unit configured to exhibit a waveform corresponding to a microphone signal that is subject to an audio interference. The audio enhancement system also includes an interference reduction unit coupled to the microphone signal and configured to provide a reduction in the audio interference, wherein a reduced audio interference is indicated by the waveform in real time. A microphone signal enhancement method is also provided.

Abstract: A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal.

Abstract: According to one embodiment, a first processing module adds, to a first queue, output sound data output from a first task, with a time stamp attached thereto. A second processing module adds, to a second queue, input sound data received from a microphone, with a time stamp attached thereto. A controller fetches first output sound data as reference data from the first queue, the first output sound data having a time stamp whose time difference from a time stamp of first input sound data in the second queue falls within a predetermined range. An echo canceller performs echo cancelling processing to cancel an echo component in the first input sound data based on the reference data.

Abstract: A method of processing a signal in a hearing instrument includes the steps of calculating a coherence between two microphone signals or microphone combination signals having different directional characteristics, determining an attenuation from the coherence, and applying the attenuation to the signal.

Abstract: The present technology provides robust, high quality dereverberation of an acoustic signal which can overcome or substantially alleviate the problems associated with the diverse and dynamic nature of the surrounding acoustic environment. The present technology utilizes acoustic signals received from a plurality of microphones to carry out a multi-faceted analysis which accurately identifies reverberation based on the correlation between the acoustic signals. Due to the spatial distance between the microphones and the variation in reflection paths present in the surrounding acoustic environment, the correlation between the acoustic signals can be used to accurately determine whether portions of one or more of the acoustic signals contain desired speech or undesired reverberation. These correlation characteristics are then used to generate signal modifications applied to one or more of the received acoustic signals to preserve speech and reduce reverberation.

Abstract: An audio apparatus is provided. The audio apparatus includes at most one electroacoustic transducer; and an audio controller, coupled to the electroacoustic transducer, for actively controlling the electroacoustic transducer to function as a loudspeaker or a microphone, wherein the loudspeaker converts output electrical signals to output sounds, and the microphone converts input sounds to input electrical signals.

Abstract: A reverberant characteristic of an acoustic space is superimposed on an audio signal that is received by an apparatus. The apparatus decomposes the audio signal into an estimated original dry signal component and an estimated reverberant characteristic of the acoustic space. Estimation of the original dry signal component and the reverberant characteristic of the acoustic space is based on determination of an estimated impulse response of the acoustic space from the received audio signal. Once the audio signal is decomposed, the estimated original dry signal component and the estimated reverberant characteristic of the acoustic space may be independently modified by the apparatus. The modified or unmodified estimated original dry signal component and estimated reverberant characteristic of the acoustic space may be combined by the apparatus to produce one or more adjusted frequency spectra.

Abstract: An acoustic echo cancellation (AEC) system includes a remote device, for capturing a remote captured sound, a server coupled to the remote device, and a local device coupled to the server. The server transmits the remote captured sound from the remote device to the local device. The local device receives, stores and plays the remote captured sound as a local playback sound. An echo is generated from reflection of the local playback sound. The local device captures the echo and a local sound into a local captured sound, and transmits both the remote captured sound and the local captured sound to the server. The server performs AEC on the local captured sound by using the remote captured sound from the local device and transmits the AEC processed local captured sound to the remote device.

Abstract: A conferencing system may comprise an electronic display configured to display remote video generated within a remote conference room, a speaker configured to reproduce remote audio generated within the remote conference room, and a processor configured to receive local audio generated within a local conference room and picked up by a microphone assembly that is part of a separate device from the electronic display. A related method may include displaying remote video on an electronic display of an all-in-one display, reproducing remote audio through at least one speaker of the all-in-one display; and performing echo cancellation of local audio using the remote audio as an echo cancellation reference. Another method may include receiving, at an all-in-one display, a plurality of local audio signals from a plurality of microphone assemblies that are separate from the all-in-one display, and controlling, within the all-in-one display, gating of the plurality of microphone assemblies.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: A signal processing system includes microphone units connected in series and a host device connected to one of the microphone units. Each of the microphone units has a microphone, a temporary storage memory, and a processing section for processing the sound picked up by the microphone. The host device has a non-volatile memory in which a sound signal processing program for the microphone units is stored. The host device transmits the sound signal processing program read from the non-volatile memory to each of the microphone units. Each of the microphone units temporarily stores the sound signal processing program in the temporary storage memory. The processing section performs a process corresponding to the sound signal processing program temporarily stored in the temporary storage memory and transmits the processed sound to the host device.

Abstract: A communication system enhances communications in a noisy environment. Multiple input arrays convert a voiced or unvoiced signal into an analog signal. A converter receives the analog signal and generates digital signals. A digital signal processor determines temporal and spatial information from the digital signals. The processed signals are then converted to audible sound.

Abstract: An input signal is processed through noise suppression (NS) and echo control (EC) via a multipath model that reduces noise pumping effects while maintaining EC performance. A copy of a “noisy” input signal is sent to an EC component before the noisy signal is sent to a NS component, which processes the signal first, when there is a consistent noise level for estimation. The copy of the pre-processing noisy signal is sent to the EC component along with a “clean” or “noise-suppressed” signal output from the NS component. The EC component analyzes the noisy signal as if the EC was the first component in the signal chain to determine what actions to take. The EC component then applies these actions to the clean signal received from the NS component.

Abstract: Embodiments of the present disclosure provide systems and methods for source localization using a microphone array. Such an embodiment, among others, utilizes circuitry to process audio signals and microphone signals to remove an echo (produced by a nearby loudspeaker) from the microphone signals using a plurality of adaptive echo cancellation filters. Location information is then determined for the loudspeaker based on the plurality of adaptive echo cancellation filters used to remove the echo.

Abstract: An input signal is supplied to a loudspeaker-room-microphone system having a transfer function and that provides an output signal. An adaptive filter unit models the transfer function of the loudspeaker-room-microphone system and provides an approximated output signal, where the output signal and the approximated output signal are subtracted from each other to provide an error signal.

Abstract: Audio input may be received at one or more microphones of a mobile device. Based on the audio input, a change in an acoustic environment of the device, such as a change in a direction of arrival of the audio input or a degradation in a quality of acoustic echo cancellation being performed upon the audio input, may be detected. A determination may be made that the detected change coincides with a non-acoustic physical event detected using an auxiliary sensor at the mobile device. The event may for example be device motion, a new proximate object, a change in a proximity of an object, a new heat source or a change in a heat level from a known heat source. Based on the determining, a signal processor, possibly comprising an audio beamformer or echo canceller, may be recalibrated, e.g. the audio beamformer or echo canceller may be caused to reconverge.

Abstract: An apparatus for canceling an acoustic echo signal caused by a far-end talker signal is provided. The apparatus for canceling an acoustic echo signal includes: a variance estimating unit configured to estimate a variance of a first audio signal of a near-end talker signal and a first noise signal of the near-end talker signal; a step size determining unit configured to determine a step size by using the variance of the first audio signal and the variance of the first noise signal; an adaptive filter coefficient updating unit configured to update an adaptive filter coefficient of an adaptive filter by using the step size; and an acoustic echo canceling unit configured to estimate an acoustic echo signal by using the adaptive filter coefficient, and cancel the acoustic echo signal from a microphone input signal by using the estimated acoustic echo signal.

Abstract: The invention provides a method for determining a signal component for reducing noise in an input signal, which comprises a noise component, comprising the steps of: estimating the noise component in the input signal, estimating a reverberation component in the noise component, and removing the estimated reverberation component from the estimated noise component to obtain a modified estimate of the noise component.

Abstract: An audio processing device for reducing the effect on a first signal of echo from a second signal, the device comprising: an echo reduction processor for processing the first signal to reduce echo in it, the echo reduction unit having: a first mode of operation for reducing echo of a first function from the first signal; and a second mode of operation for reducing echo of a second function from the first signal, the second function being more complex than the first function and the echo reduction processor being such as to consume more power in the second mode of operation than in the first mode of operation; and an echo reduction controller for controlling the echo reduction processor to operate in a selected one of the first mode of operation and the second mode of operation.

Abstract: Systems and methods for providing karaoke recording and playback on mobile devices are provided. The mobile device may play music audio and associated video, and receive via one or more microphones a mix of a user voice, the music, and background noise. The mix is stored both in its original form and as processed to enhance voice and sound through noise suppression and other processing. Stored audio may be uploaded through a communications network to a cloud based computing environment for listening on other mobile devices. Selectable playing control and recording options may be provided. Audio cues may be determined during signal processing of the original acoustic sound and be stored on the mobile device. During playback of recorded audio and, optionally, associated video, the original acoustic sound, recorded cues, and user selectable optional processing may be used to remix during playback, while retaining the original recording.

Abstract: The present invention discloses an echo cancellation method. The method includes: dividing an audio signal into a high-band audio signal and a low-band audio signal; performing adaptive filtering on the low-band audio signal, and performing synthesis filtering on a signal obtained after the low-band audio signal undergoes the adaptive filtering and on the high-band audio signal to generate a preliminary echo cancellation signal; performing envelope predication echo suppression on a high-band signal in the preliminary echo cancellation signal, and calculating and outputting a residual echo suppression coefficient; performing echo suppression on a low-band signal in the preliminary echo cancellation signal, and outputting a processing result; and multiplying the output result by the residual echo suppression coefficient, and outputting a signal of which echoes are canceled.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: Techniques to reduce distortion in acoustic signals in mobile computing devices are described. For example, a mobile computing device may comprise a speaker operative to receive a first signal and output a second signal. The mobile computing device may further comprise a first microphone operative to receive the second signal and a second microphone operative to receive a third signal. An echo canceller may be coupled to the first microphone and the second microphone and may be operative to compare the second signal and the third signal and reduce distortion in the third signal based on the comparison. Other embodiments are described and claimed.

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: The present invention provides an audio processing device that appropriately suppresses echo generated in a stereophonic audio output. The audio processing device includes: means for generating a first artificial linear echo signal and a second artificial linear echo signal that are estimated to be generated by first audio and second audio travelling to audio input means; means for suppressing a linear echo signal mixed to an input audio signal based on the first artificial linear echo signal and the second artificial linear echo signal: means for estimating a non-linear echo signal based on the first artificial linear echo signal and the second artificial linear echo signal; and means for suppressing the non-linear echo signal.

Abstract: A new acoustic echo suppressor and method for acoustic echo suppression is described herein. Exemplary embodiments of the acoustic echo suppressor use one linear regression model for each subband. The linear regression model for each subband may operate on the squared magnitude of the input samples as well as corresponding cross-products. In this way, accurate and robust estimates of the echo signal in each subband can be obtained, thereby providing good echo reduction while keeping the signal distortion low.

Abstract: A method for operating an electronic device including a first microphone and a second microphone to block the echo generation the electronic device is operated in a loudspeaker mode. The method includes: determining whether a sensor located within a predetermined range from the second microphone has sensed a peripheral object; blocking an operation of the second microphone when the sensor has sensed a peripheral object; and blocking echo generation by eliminating a sound that is output from a speaker and input to the first microphone.

Abstract: An device includes an audio output unit, a first audio input unit, a second audio input unit that are disposed in a position closer to the audio output unit than the first audio input unit, a unit for outputting a combined signal of which audio signals from the first and second audio input unit are combined so as forming directivity in which sensitivity in a direction of the audio output unit is low when viewed from the first and second audio input units, a unit for generating artificial echo corresponding to an echo component mixed in the audio inputted to the first audio input unit, and a unit for performing an echo suppression process to the combined signal by using the artificial echo.

Abstract: A method comprises processing M subband communication signals and N target-cancelled signals in each subband with a set of beamformer coefficients to obtain an inverse target-cancelled covariance matrix of order N in each band; using a target absence signal to obtain an initial estimate of the noise power in a beamformer output signal averaged over recent frames with target absence in each subband; multiplying the initial noise estimate with a noise correction factor to obtain a refined estimate of the power of the beamformer output noise signal component in each subband; processing the refined estimate with the magnitude of the beamformer output to obtain a postfilter gain value in each subband; processing the beamformer output signal with the postfilter gain value to obtain a postfilter output signal in each subband; and processing the postfilter output subband signals to obtain an enhanced beamformed output signal.

Abstract: A direct sound extraction device includes: a spectrum transform unit that transforms an input signal, which includes a reverberant sound in a direct sound and on which a Fourier transform process has been performed, to a first amplitude spectrum signal Lfa; a low-pass filter unit (10) that performs a low-pass filtering process on the first amplitude spectrum signal Lfa for each frequency to generate a second amplitude spectrum signal Lfa1; a first subtraction unit (18) that calculates a third amplitude spectrum signal by subtracting the second amplitude spectrum signal Lfa1 from the first amplitude spectrum signal Lfa; and an inverse Fourier transform unit that generates a direct sound signal Lfd from a frequency spectrum signal calculated based on a phase spectrum signal and the third amplitude spectrum signal.

Abstract: An echo canceler 10 generates an echo elimination signal by filtering through adaptive filters 101 and 102 reference signals input from sound sources causing echoes. It includes a sound source number detecting unit 103 for detecting the number of the sound sources causing echoes from the reference signals, and a control unit 105 for making the number of taps of the adaptive filters 101 and 102 variable in accordance with the number of the sound sources detected by the sound source number detecting unit 103.

Abstract: Traditionally, echo cancellation has employed linear adaptive filters to cancel echoes in a two way communication system. The rate of adaptation is often dynamic and varies over time. Disclosed are novel rates of adaptation that perform well in the presence of background noise, during double talk and with echo path changes. Additionally, the echo or residual echo can further be suppressed with non-linear processing performed using joint frequency-time domain processing.

Abstract: An audio processing system comprising two or more microphones and an echo cancellation system configured to apply a fast converging adaptive filtering algorithm to low frequency bands of a first microphone signal to generate first synthesized echo signal components and an adaptive filtering algorithm to high frequency bands of the first microphone signal to generate second synthesized echo signal components and to apply the first synthesized echo signal components and the second synthesized echo signal components to the first microphone signal to cancel an echo signal of the first microphone signal. An echo estimate and suppression system is configured to receive the first synthesized echo signal components and the second synthesized echo signal components and to apply them to estimate powers of echo signals in one or more additional microphones.

Abstract: An audio signal processing system and an echo signal removing method thereof are provided. The audio signal processing system includes a speaker that is configured to output an audio signal; a microphone that is configured to receive the audio signal output by the speaker including an echo signal generated by the audio signal; an echo signal delay unit that is configured to delay the echo signal for a bulk delay time, and output the echo signal that is delayed; and an echo signal removing unit that is configured to remove the echo signal that is delayed and output by the echo signal delay unit from the audio signal received by the microphone, wherein the echo signal delay unit includes a bulk delay measuring unit that is configured to measure a bulk delay by analyzing impulse response characteristics of an echo path.

Abstract: A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.

Abstract: According to one embodiment, information terminal includes: first audio input module; first audio output module; second audio input module; audio processor; and second audio output module. The first audio input module receives an external sound from an external information terminal connected to the information terminal via an external network. The first audio output module outputs the external sound. The second audio input module receives sounds transmitted from a sound input device of each information terminal within a group connected to the information terminal via the internal network. The audio processor synthesizes the in-group sounds to generate a single input sound, and removes an echo component from the input sound. Here, the echo component is caused due to the external sounds output from the sound output device. The second audio output module outputs the input sound from which the echo component is removed to the external information terminal.