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: Audio signals are processed for use in a communication event. A data store may be queried to obtain an indication of an echo direction, which relates to a direction from which audio signals output from the audio output are likely to be received at a microphone array (plurality of microphones) of a device. Beamformer coefficients of an adaptive beamformer of the device are determined in dependence upon the received indication of the echo direction. Audio signals are received at the microphone array. The adaptive beamformer applies the determined beamformer coefficients to the received audio signals, thereby generating a beamformer output for use in the communication event. The beamformer coefficients are determined such that echo suppression is applied to audio signals received at the microphone array from the indicated echo direction.

Abstract: A wideband acoustic echo cancellation apparatus with an adaptive tail length in an embedded system, and a wideband acoustic echo cancellation method are provided, and the wideband acoustic echo cancellation apparatus may include a delay length calculating unit to calculate a delay length of an echo path, using a near-end signal and a far-end signal, an adaptive filter implementing unit to implement an adaptive filter based on the calculated delay length, using selected coefficients, and an error calculating unit to search for three intervals having a largest impulse response value from all intervals of a tail of the adaptive filter, and to calculate an error during an interval in which the selected coefficients are used.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include identifying a target sub-region of the frequency response which has a predetermined area percentage of a total area under a curve generated by the frequency response, determining whether the target sub-region is a narrow energy region, creating a filter to adjust the frequency response, and applying the filter to the audio signal.

Abstract: A system that utilizes closed-form solutions to perform echo cancellation is described. The system includes a filter, filter parameter determination logic and a combiner. The filter is configured to process a far-end audio signal in accordance with one or more filter parameters to generate an estimated echo signal. The filter parameter determination logic is configured to update estimated statistics associated with the far-end audio signal and a microphone signal based on instantaneous statistics associated with the far-end audio signal and the microphone signal, and calculate the one or more filter parameters based upon the updated estimated statistics. The combiner is configured to generate an estimated near-end audio signal by subtracting the estimated echo signal from the microphone signal.

Abstract: A variable-bandwidth adaptive notch filter which cancels howling from an input signal with a bandwidth varying according to a howling frequency to generate an output signal.

Abstract: Systems and methods for performing videoconferencing using split endpoints are disclosed herein. These endpoints can include a Main Unit and a Satellite Unit that are associated with each other, and where the tasks of video and audio encoding and decoding are separated between the two. In some embodiments the Main Unit performs video and audio encoding, and the Satellite Unit performs video and audio decoding. In some embodiments the Satellite Unit obtains media data through its associated Main Unit, whereas in other embodiments the Satellite Unit obtains media data directly from the Server.

Abstract: The invention is directed to a single channel mask estimation method capable of improving reverberant speech identification for CI users. The method is based on the energy of the reverberant signal and the residual signal computed from linear prediction (LP) analysis. The mask is estimated by comparing the energy ratio of the two signals at different frequency bins with an adaptive threshold. As the threshold is updated for each frame of speech based on the energy ratios of the reverberant and LP residual signals computed from previous frames, it is amenable for real-time implementation. It can thus be used as a specialized (for reverberant environments) sound coding strategy used for cochlear implant applications.

Abstract: Provided are a method and device for eliminating echo. The method comprises: an echo path characteristic parameter of an echo signal is estimated; a source signal of the echo signal is taken as a reference signal, and an echo estimation signal is generated according to the echo path characteristic parameter; and the echo estimation signal is subtracted from a speech signal to be processed. The disclosure solves the problem in the related art that self-adaptation cannot be achieved when a returned near-end audio is suppressed, thus facilitating the enhancement of the speech quality of a conference and the improvement of user experience.

Abstract: There is provided a sampling point adjustment apparatus including: a frequency conversion unit that converts a first signal and a second signal into a first narrow band signal and a second narrow band signal through frequency conversion; a central position determination unit that determines a central position of a window of the frequency conversion for the second signal based on an estimated value of a sampling interval offset between the first narrow band signal and the second narrow band signal; and a phase control unit that controls a phase of the second narrow band signal based on the estimated value.

Abstract: Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker.

Abstract: A processing system can include a tracking microphone array; audio tracker circuitry connected to the tracking microphone array to track an audio source based on an audio input from the array; communication microphones; and a processor. The processor can include audio circuitry to receive an audio input from the communication microphones and process the audio input to apply one or more of acoustic echo cancellation (AEC) and acoustic echo suppression (AES) processing to the audio input. The processor can further include calculating circuitry to calculate a ratio of signal power after and before the AEC and/or the AES processing, and control circuitry to generate an acoustic echo presence indication based on the ratio calculated by the calculating circuitry. The processor can transmit, via transmitting circuitry, the acoustic echo presence indication to an audio tracking device via a data communication channel between the processor and the audio tracker.

Abstract: The performance of an echo canceller is assessed using a) a test signal launched from originating test equipment and b) a simulated echo of the test signal launched from terminating test equipment. The launch of the simulated echo signal is timed in such a way that it arrives at the tandem echo canceller(s) at a particular point in time relative to the arrival of the test signal, at the tandem echo canceller(s), when the tandem echo canceller(s) is (are) not able to cancel the simulated echo signal. The latter thus arrives uncanceled at the target echo canceller. The launch of the simulated echo signal is further timed in such a way that it arrives at the target echo canceller at a point in time relative to the arrival of the test signal, at the target echo canceller, when the target echo canceller is able to cancel the simulated echo signal.

Abstract: Generally, this disclosure provides devices, systems and methods for cancelling an interfering audio signal. The system may include a mobile device including a microphone configured to capture an acoustic audio signal, the acoustic audio signal a combination of the interfering audio signal and a desired audio signal, the desired audio signal generated by a user of the mobile device. The system may also include a wireless communication module incorporated in the mobile device, to receive a reference signal through a side-channel, the reference signal associated with the interfering audio signal. The system may further include an acoustic echo cancellation module incorporated in the mobile device, the acoustic echo cancellation module to cancel the interfering audio signal from the captured acoustic audio signal, the cancellation based on the reference signal.

Abstract: A reverberation suppressing apparatus, includes: a sound acquiring unit which acquires a sound signal; a reverberation data computing unit which computes reverberation data from the acquired sound signal; a reverberation characteristics estimating unit which estimates reverberation characteristics based on the computed reverberation data; a filter length estimating unit which estimates a filter length of a filter which is used to suppress a reverberation based on the estimated reverberation characteristics; and a reverberation suppressing unit which suppresses the reverberation based on the estimated filter length.

Abstract: A system and method of matching reverberation in teleconferencing environments. When the two ends of a conversation are in environments with differing reverberations, the method filters the reverberation so that when both signals are output at the near end (e.g., the audio signal from the far end and the sidetone from the near end), the reverberations match. In this manner, the user does not perceive an annoying difference in reverberations, and the user experience is improved.

Abstract: A method and system for acoustic echo cancellation varies a step size of an adaptive filter in an acoustic echo canceller. Far-end data is received and echo estimate data is calculated using the received far-end data. Microphone data is received and error data is calculated using the received microphone data and the echo estimate data. A first average of the microphone data and a second average of the error data are computed over a predefined number of samples. An echo leakage is estimated using the first average and the second average wherein the echo leakage indicates an extent to which the far-end data is present in the error data, and the step size of the adaptive filter is varied based on the echo leakage and a maximum allowed step size.

Abstract: This document discloses one or more systems, apparatuses, methods, etc. for implementing an echo cancellation algorithm for long delayed echo that is created during a wire or wireless voice communications. In an implementation, a WiDi feature in a device during the wire or wireless voice communications may add an additional echo delay in addition to channel multipath delay when an audio sound signal travels from a WiDi component to a microphone of the device. In this implementation, a separate delay estimator is configured to estimate total delay. The estimated total delay is fed back to an adoptive filter component for long delay echo cancellation.

Abstract: A microphone module disposed in an electronic device for reducing echo noise. The microphone module includes a casing, a first diaphragm disposed in the casing, a second diaphragm disposed in the casing and a substrate disposed between the first diaphragm and the second diaphragm and joined to the casing to define a first space and a second space which are isolated and separated from each other. The first diaphragm is disposed in the first space, the second diaphragm is disposed in the second space, and the substrate is electrically connected with the first diaphragm and the second diaphragm.

Abstract: A method and system for acoustic echo cancellation stores received far-end data in a first buffer. When the far-end data in the first buffer exceeds a predefined length, the stored far-end data is used to calculate echo estimate data. The echo estimate data is stored in a second buffer. Whenever microphone data is received the error data is calculated independent of echo estimate data availability. In particular, subsequent to sufficient echo estimate data being stored in the second buffer and responsive to the reception of the microphone data, the error data is calculated by subtracting, from the microphone data, corresponding echo estimate data stored in the second buffer.

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: Acoustic echo cancellation is improved by receiving a speaker signal that is used to produce audio in a room, and receiving audio signals that capture audio from an array of microphones in the room, including an acoustic echo from the speakers. To cancel the acoustic echo, one adaptive filter is associated with a corresponding subspace in the room. Each of the audio signals is assigned to at least one of the adaptive filters, and a set of coefficients is iteratively determined for each of the adaptive filters. The coefficients for an adaptive filter are determined by selecting each of the audio signals assigned to that adaptive filter and adapting the filter to remove an acoustic echo from each of the selected audio signals. At each iteration, a different audio signal is selected from the audio signals assigned to the adaptive filter in order to determine the set of coefficients.

Abstract: A controller for an audio device is provided. The controller receives a first collected sound signal and a second collected sound signal respectively provided by two microphones, and includes an echo cancellation module and a beamforming module. The echo cancellation module performs echo cancellation on the first collected sound signal to accordingly provide an intermediate signal. The beamforming module performs beamforming by utilizing the echo-cancelled intermediate signal and the non-echo-cancelled second collected sound signal.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: The present disclosure regards a hearing device comprising a power source, electric circuitry, a loudspeaker, at least one microphone for sound from an acoustic environment, and at least one wireless receiver for wirelessly received sound signals. The microphone is configured to generate an environment sound signal. The wireless receiver is configured generate a source sound signal. The electric circuitry is configured to estimate at least one parameter of an impulse response from the location of the origin of the wirelessly received signal to the location of a user of the hearing device in dependence on the source sound signal and the environment sound signal. The electric circuitry is further configured to process the environment sound signal in dependence on the estimated at least one impulse-response parameter, thereby generating an output sound signal. The output sound signal is processed into sound by the loudspeaker.

Abstract: The present invention relates to an earset, comprising: a first earphone portion, which includes a first speaker for outputting sound signals or voice signals that are provided from an external device, and which can be inserted into a first external auditory canal of a user; a second earphone portion, which includes a first microphone for receiving inputted user voice signals that are provided through the external auditory canal of the user, and which can be inserted into a second auditory canal of the user; and a main body connected to each of the first and the second earphone portion. When the main body is wirelessly connected to the external device, the main body comprises; a signal transceiving portion for transceiving the signals with the external device; and a control portion for outputting via the first speaker the voice signals received from the external device through the signal transceiving portion.

Abstract: A method using an may microphone to cancel echo applies to a sound receiving system and comprises steps: an array microphone receiving a sound source and outputting a plurality of analog acoustic signals formed from the sound source; an A/D converter converting the analog acoustic signals into a plurality of digital acoustic signals; a digital signal processor respectively using an adaptive beamforming process and a blocking matrix filtering process to convert the digital acoustic signals into a primary acoustic signal and at least one noise signal; and the digital signal processor using a multiple-input cancelling process to subtract the noise signal from the primary acoustic signal to obtain an acoustic signal where the echo has been cancelled. Thereby, the present invention can eliminate the systematic errors of the array microphone of the sound receiving system and improves the robustness of the acoustic signal.

Abstract: Provided is an acoustic signal processing device for producing an output sound meeting listener's preferences by adjusting attack sound, reverberation, and noise component.

Abstract: Near-end equipment for a communication channel with far-end equipment. The near-end equipment includes at least one loudspeaker, at least two microphones, a beamformer, and an echo canceller. The communication channel may be in one of a number of communication states including Near-End Only state, Far-End Only state, and Double-Talk state. In one embodiment, when the echo canceller determines that the communication channel is in either the Far-End Only state or the Double-Talk state, the beamformer is configured to generate a nearfield beampattern signal that directs a null towards a loudspeaker. When the echo canceller detects the Near-End Only state, the beamformer is configured to generate a farfield beampattern signal that optimizes reception of acoustic signals from the near-end audio source. Using different beamformer processing for different communication states allows echo cancellation processing to be more successful at reducing echo in the signal transmitted to the far-end equipment.

Abstract: Embodiments of method and apparatus for acoustic echo control are described. According to the method, an echo energy-based doubletalk detection is performed to determine whether there is a doubletalk in a microphone signal with reference to a loudspeaker signal. A spectral similarity between spectra of the microphone signal and the loudspeaker signal is calculated. It is determined that there is no doubletalk in the microphone signal if the spectral similarity is higher than a threshold level. Adaption of an adaptive filter for applying acoustic echo cancellation or acoustic echo suppression on the microphone signal is enabled if it is determined that there is no doubletalk in the microphone signal through the echo energy-based doubletalk detection, or there is no doubletalk through the spectral similarity-based doubletalk detection.

Abstract: In a conventional dereverberation system, when there is a fluctuating reverberation component, it has been difficult to determine, with high accuracy, a linear dereverberation filter for removing a non-fluctuating reverberation component. An algorithm integrating a dereverberation system using a linear filter and a dereverberation system using a non-linear filter includes the function of measuring the amount of fluctuation in transfer function in a latter-stage non-linear filter over time, and controls the strength of the non-linear filter over time based on the function. In this configuration, a strong non-linear process is implemented only when the fluctuation in transfer function is large, whereby distortion in speech components can be minimized.

Abstract: Method for creating an audio environment having N speakers HPi, i=1 . . . N fed by N signals Si, i=1 . . . N generated from M theoretical signals STj, j=1 . . . M provided to feed M theoretical speakers HPTj, j=1 . . . M , wherein: position information is determined relating to the N speakers HPi, i=1 . . . N and a listening point, the two theoretical speakers HPTj and HPTj+1 which would be angularly closest to a speaker HPi, the signal Si is determined according to the following equation: Si=Gi[STj(GpijGeij)+STj+1(Gpi(j+1)Gei(j+1))]e?i??i wherein: Gpij and Gpi(j+1) are panning gains, Geij and Gei(j+1) are balancing gains Gi and i are a positioning gain and delay, respectively, which enable the speakers HPi, i=1 . . . N to be virtually repositioned in terms of distance so that all sounds intended to simultaneously arrive at the listening point according to the encoding format actually arrive therein simultaneously, irrespective of the remoteness of the speakers relative to the listening point.

Abstract: Methods and systems are provided for acoustic echo cancellation in electronic devices. The echo cancellation may comprise applying, as a first step, echo cancellation filtering to an acoustic input obtained via an acoustic input element (e.g., microphone), and applying, as a second step, echo suppression to the acoustic input, wherein the echo suppression comprises suppressing residual echo in the acoustic input. The echo cancellation filtering may comprise identifying and/or filtering out echo components, both linear and nonlinear, in the acoustic input, with the echo components corresponding to an echo signal caused by an acoustic output outputted via the acoustic output element (e.g., speaker). A sensor signal, generated by a vibration sensor that detects vibrations in the electronic device including vibrations caused by the outputting of the acoustic output, may be used as reference signal in the echo cancellation filtering and/or the echo suppression.

Abstract: An encoding device, a decoding device, and related methods are provided that eliminate the loss of synchronization of the adaptive filters of a terminal at the encoding end and a terminal at the decoding end caused by transmission errors. Deterioration of the sound quality is suppressed when a multiple channel signal is encoded with high efficiency using an adaptive filter. In the terminal at the encoding end, a buffer stores updated filter coefficients. When packet loss detection information indicating whether there is any packet loss in the terminal at the decoding end indicates that there is packet loss, a switch outputs the past filter coefficients of the previous (NX+1) frames from the buffer to an adaptive filter. The adaptive filter uses the past filter coefficients of the previous (NX+1) frames to conduct filtering.

Abstract: Provided are a method for estimating a spectrum density of diffused noises. Also provided is a processor for implementing the method. The processor includes at least two sound receiving units and a spectrum density estimating unit for estimating spectrum density.

Abstract: An embodiment provides a method, including: accessing a tuned corrective mask stored in a memory device; forming, using a processor, a tuned acoustic echo cancellation mask utilizing the tuned-corrective mask; applying, using a processor, the tuned acoustic echo cancellation mask to a digital audio signal; and outputting an echo-cancelled audio signal. Other aspects are described and claimed.

Abstract: Technologies are generally described for a system for controlling an audio signal based on a proximity of, e.g., a user's hand to at least one component of an audio control system. In some examples, an audio control system may include a filter configured to provide an echo signal and a control decision unit configured to provide a control signal based on the echo signal.

Abstract: A method for multi-channel echo cancellation and noise suppression is described. One of multiple echo estimates is selected for non-linear echo cancellation. Echo notch masking is performed on a noise-suppressed signal based on an echo direction of arrival (DOA) to produce an echo-suppressed signal. Non-linear echo cancellation is performed on the echo-suppressed signal based, at least in part, on the selected echo estimate.

Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.

Abstract: An acoustic echo suppression unit according to an embodiment of the present invention includes and input interface for extracting a downmix signal from an input signal, the input signal including the downmix signal and parametric side information, wherein the downmix and the parametric side information together represent a multichannel signal, a calculator for calculating filter coefficients for an adaptive filter, wherein the calculator is adapted to determine the filter coefficients based on the downmix signal and a microphone signal or a signal derived from the microphone signal, and an adaptive filter adapted to filter the microphone signal or the signal derived from the microphone signal based on the filter coefficients to suppress an echo caused by the multichannel signal in the microphone signal.

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: A dereverberation apparatus includes a signal selecting unit which selects a sound signal to be used for dereverberation process from a plurality of sound signals, and a dereverberation processing unit which performs the dereverberation process for the selected sound signal.

Abstract: Methods and apparatus for beamforming and performing echo compensation for the beamformed signal with an echo canceller including calculating a set of filter coefficients as an estimate for a new steering direction without a complete adaptation of the echo canceller.

Abstract: Methods, systems, and apparatuses are described for performing acoustic echo cancellation with internal upmixing that allow for a more effective handling of acoustic echo cancellation of audio components that are provided via different channels. In an embodiment in which audio is played back using two loudspeakers, audio components that are panned equally among the loudspeakers form a “phantom center image.” Acoustic echo cancellation is performed by initially upmixing the different channels to internally create modified versions of these channels and a virtual channel representative of the phantom center image. Each of these channels is passed through a respective adaptive filter that is configured to estimate an acoustic echo produced by each respective channel. These estimates are then subtracted from the signal received from one or more microphones (or from a signal obtained by combining multiple microphone signals) to suppress or eliminate the acoustic echo.

Abstract: A microphone system is provided, wherein the microphone system comprises a microphone array comprising a plurality of microphone units each adapted to generate a primary signal indicative of an acoustic wave received from the respective microphone unit, a first echo cancellation unit, an integrator unit, and a combination unit, wherein the microphone system is adapted to generate a first dipole response and a monopole response from the primary signals, wherein the integrator unit is adapted to generate a first integrated dipole response by integrating the first dipole response, wherein the first echo cancellation unit is adapted to generate a first echo cancelled integrated dipole response from the first integrated dipole response, and wherein the combination unit is adapted to combine the monopole response and the first echo cancelled integrated dipole response.

Abstract: Provided is a sound emission and collection device capable of estimating the azimuth of a sound source (such as a main utterer) precisely without any processing load. The sound emission and collection device (1) is connected with another sound emission and collection device via a network or the like. The sound emission and collection device (1) receives a sound signal from another sound emission and collection device, as a sound emission signal (FE), and emits the same from a speaker (SP). The sound emission and collection device (1) collects the sound at microphones (MIC1 to MIC3), and produces sound collection beam signals (NE1 to NE3) of different azimuths. The sound emission and collection device down-samples the individual sound collection beam signals (NE1 to NE3), and filters out the echoes of the down-sampled sound collection beam signals (DNE1 to DNE3).

Abstract: The initial values of parameter estimates are set, including reverberation parameter estimates, which includes a regression coefficient used in a linear convolutional operation for calculating an estimated value of reverberation included in an observed signal, source parameter estimates, which includes estimated values of a linear prediction coefficient and a prediction residual power that identify the power spectrum of a source signal, and noise parameter estimates, which include noise power spectrum estimates. Then, the maximum likelihood estimation is used to alternately repeat processing for updating at least one of the reverberation parameter estimates and the noise parameter estimates and processing for updating the source parameter estimates until a predetermined termination condition is satisfied.

Abstract: Method, user terminal and computer program product for controlling audio signals at the user device during a communication session between the user device and a remote node, in which a primary audio signal is received at audio input means of the user device for transmission to the remote node in the communication session. It is determined whether the user device is operating in (i) a first mode in which secondary audio signals output from the user device are likely to disturb the primary audio signal received at the audio input means, or (ii) a second mode in which secondary audio signals output from the user device are not likely to disturb the primary audio signal received at the audio input means.

Abstract: An echo cancelling device splits a low-band signal through LPFs having characteristics which do not cause aliasing during downsampling of downsamplers, and splits a high-band signal through HPFs having characteristics which do not cause aliasing during downsampling of downsamplers. The echo canceling device generates a mid-band signal by subtracting the low-band signal and the high-band signal from a pre-split signal by adder-subtractors, and cancels an echo on a band-by-band basis.

Abstract: Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.