Abstract: Provided is an acoustic reproduction device that includes a first acoustic reproduction unit, and a second acoustic reproduction unit, in which the first acoustic reproduction unit includes a housing having a cylindrical shape and a vibration exciter that vibrates an end surface of one end of the housing. The second acoustic reproduction unit includes a speaker unit and a diffuser that changes a radiation direction of sound reproduced by the speaker unit. The housing, the speaker unit, and the diffuser are arranged so as to be substantially coaxial with a predetermined axis, and the diffuser causes a radiation direction of sound reproduced by the speaker unit and a radiation direction of sound from the first acoustic reproduction unit to be substantially a same.

Abstract: A method for filtering noise of lens operation is applicable to a controller being communicably connected with a video recorder. The method includes: sending, by the controller, a recording instruction configured to control the video recorder to perform a video recording and an audio recording; sending, by the controller, a lens operation instruction configured to control a lens of the video recorder to perform an operation; reading, by the controller, an inverted noise signal associated with the operation receiving, by the controller, an audio recording signal generated by the video recorder when the video recorder performs the operation; and synthesizing, by the controller, the audio recording signal and the inverted noise signal to output a synthesized audio signal.

Abstract: Devices, methods, and systems for secure communications on a computing device. A host operating system (OS) runs on a host processor in communication with a host memory. A secure OS runs on a coprocessor in communication with a secure memory. The coprocessor receives information from an external device over a secure peer-to-peer (P2P) connection. The secure P2P connection is managed by the secure OS and is not accessible by the host OS.

Abstract: In an audiovisual system, in which video is displayed on a screen that does not permit sound to pass through the screen, such as a light emitting diode panel, a high-frequency speaker positioned above an audience seating area can direct sound toward the screen, so that the screen can reflect the sound toward the audience seating area. The high-frequency speaker can be used with one or more low-frequency speakers positioned at or near the height of the audience seating area. The low-frequency and high-frequency sounds can appear to originate from close to the same height, thereby creating a realistic audio image at the audience seating area. A spectral filter can negate the spectral effects of propagation to and reflection from the screen. Suitable time delays can synchronize the high-frequency sound with the low-frequency sound and with video displayed on the screen.

Abstract: Systems and methods for audio communication are disclosed. The system includes a plurality of transducer units and plurality of sensors adapted to be located in a plurality of sites/spaces to which service should be provided by the system. The plurality of transducer units are capable of emitting/directing and focusing ultra-sonic signals to respective coverage zones in the sites, such that localized (confined) sound field can be formed at selected spatial position in the coverage zones by utilizing sound from ultrasound technique. The sensors are associated with respective sensing volumes in the sites and are operable to obtain sensory data indicative of spatial arrangement of elements/object at the sensing volumes within the sites.

Abstract: A frequency compressor of a vibration output apparatus generates a compressed signal by converting frequency components of a mid-band signal into low-band frequency components of a low-band signal by increasing the total number of samples by a factor of n and thus compressing the frequency of amplitude information included in mid-band frequency components of the mid-band signal to 1/n. If the level of a low-band envelope signal is lower than a predetermined threshold level, a vibration signal generator generates a vibration signal by combining the compressed signal with the low-band signal consisting of the low-band frequency components of an acoustic signal. If the level of the low-band envelope signal is higher than the predetermined threshold level, the vibration signal generator generates a vibration signal by directly using the low-band signal. A vibration output unit outputs a vibration on the basis of the generated vibration signal.

Abstract: A motion measurement apparatus according to an embodiment of the present technology includes a detector unit, a controller unit, and an output unit. The detector unit is to be attached to a detection target and detects velocity-related information, the velocity-related information being related to temporal changes in velocities in three-axis directions of the detection target being in motion in a space. The controller unit extracts a motion feature amount including one or a plurality of kinematic physical quantities of the detection target on the basis of the velocity-related information. The output unit generates a perceivable measurement signal that changes depending on the motion feature amount.

Abstract: A sound producing device is disclosed. The sound producing device includes a first air pulse generating element, driven by a first driving signal comprising a first electrical pulse, the first electrical pulse comprises a first transition edge with a first transition polarity; and a second air pulse generating element, disposed adjacent to the first air pulse generating element, driven by a second driving signal comprising a second electrical pulse, the second electrical pulse comprises a second transition edge with a second transition polarity; wherein the first transition polarity is opposite to the second transition polarity; wherein the first transition edge generally coincides with the second transition edge; wherein air pulses generated by the first or the second air pulse generating element produce non-zero offset in terms of sound pressure level.

Abstract: An air pulse generating element is disclosed. The air pulse generating element includes a membrane, actuated according to a driving signal; a plate, wherein a chamber is formed between the plate and the membrane; wherein during a first phase of the driving signal, the membrane is actuated to change a volume within the chamber to generate an air pulse; wherein during a second phase of the driving signal, the membrane is actuated such that a gap is temporarily formed; wherein the temporarily formed gap is configured to provide a temporary air shunt to accelerate an air balancing process between a first air pressure at a first side of the membrane and a second air pressure at a second side of the membrane.

Abstract: Disclosed are method and device for outputting audio linked with a video screen zoom. The method of the present invention comprises the steps of: receiving an audio signal linked with a video and generating a decoded audio signal and object metadata; linking with an execution of a video screen zoom so as to change respective object gain values for objects which are present in a zoom area; and rendering the decoded audio signal by using the changed object gain values. In addition, the method for outputting audio, of the present invention, further comprises a step for changing a channel signal-specific gain value which is linked with the video screen zoom. An audio can be linked, through the method and device for outputting audio, according to an embodiment of the present invention, in accordance with the video screen zoom such that the audio which is more realistic can be provided.

Abstract: A method is provided that can be performed by an audio source for negotiating latency in an audio network. The audio source receives information regarding an audio processing latency associated with each of a set of two or more audio output devices connected to the audio network. The audio source determines, based at least on the received information, a maximum delay for outputting audio samples streamed by the audio source from the audio output devices in the set. The audio source determines, based on the maximum delay, timing for outputting the audio samples from the audio output devices in the set. The audio source then communicates the determined timing to the set of audio output devices for processing the audio samples in accordance with the timing.

Abstract: Disclosed is a sound data processing system including a plurality of sound collecting terminals and a server. Each of the plurality of sound collecting terminals includes: a sound collecting means that collects a sound and has an orientation direction, and a communication module that transmits sound data corresponding to the sound collected by the sound collecting means and supplementary data including position data of a position of a corresponding sound collecting terminal and orientation direction data corresponding to the orientation direction of the sound collecting means via a network. The server receives the sound data and the supplementary data transmitted by the plurality of sound collecting terminals through the network and determines a position of a source that emits the sound collected by the sound collecting means on the basis of the sound data and the supplementary data.

Abstract: A sound producing device (SPD) is provided. The SPD comprises a membrane, having a resonance frequency and a resonance bandwidth; and an actuator, disposed on the membrane, receiving a driving signal corresponding to an input audio signal; wherein the input audio signal has an input audio band which is upper bounded by a maximum frequency; wherein the resonance frequency is higher than the maximum frequency plus a half of the resonance bandwidth.

Abstract: Offloading of location computation from a location server to an access point through the use of projections on base phase vectors may be provided. First, an Access Point (AP) may receive a set of two or more base phase vectors from a location server. Next, the AP may measure a measured phase vector for a first signal from a user device. Then, the AP can determine projection values based on a comparison of the measured phase vector to each base phase vector. From these comparisons, the AP can determine a subset of base phase vectors with the highest projection values. The AP can then send the projection values and the subset of base phase vectors to the location server, wherein the location server determines the device location from these projection values and subset of base phase vectors.

Abstract: A system may include at least one audio transducer and a controller. The controller may generate at least one actuation signal. The at least one actuation signal may drive the at least one audio transducer such that the at least one audio transducer generates evanescent wave audio signals in proximity to a wearer's ear. The evanescent wave audio signals may decay in strength with distance from the at least one audio transducer. Various other apparatuses, systems, and methods are also disclosed.

Abstract: The present technology relates to a signal processing device, a signal processing method, and a program that enable different sounds to be reproduced in a remote location and a neighboring location. A signal processing device includes: a remote filter unit configured to generate a remote sound reproduction signal for reproducing a sound in a remote audible region, by performing filter processing on a first sound source signal using a remote sound reproduction filter coefficient; and a neighboring filter unit configured to generate a neighboring sound reproduction signal for reproducing a sound in a neighboring audible region that is different from the remote audible region, by performing filter processing on a second sound source signal using a neighboring sound reproduction filter coefficient. The present technology can be applied to a remote-neighborhood separate sound field formation device.

Abstract: A microphone array includes one or more front-facing microphones disposed on a front surface of the collaboration endpoint and a plurality of secondary microphones disposed on a second surface of the collaboration endpoint. The sound signals received at each of the one or more front-facing microphones and the plurality of secondary microphones are converted into microphone signals. When the sound signals have a frequency below a threshold frequency, an output signal is generated from microphone signals generated by the one or more front-facing microphones and the plurality of secondary microphones. When the sound signals have a frequency at or above a threshold frequency, an output signal is generated from microphone signals generated by only the one or more front-facing microphones.

Abstract: An air pulse generating element, disposed in a sound producing device, includes a membrane, disposed within a chamber; and a plurality of valves, disposed by the membrane within the chamber, configured to seal a plurality of openings of the chamber in response to a plurality of valve control signals; wherein the membrane and the plurality of valves are all fabricated at a first layer.

Abstract: An air pulse generating element, disposed in a sound producing device, includes a membrane, disposed within a chamber; and a plurality of valves, disposed by the membrane within the chamber, configured to seal a plurality of openings of the chamber in response to a plurality of valve control signals; wherein the membrane and the plurality of valves are all fabricated at a first layer.

Abstract: A three-dimensional (3D) audio reproducing method and apparatus is provided. The 3D audio reproducing method may include receiving a multichannel signal comprising a plurality of input channels; and performing downmixing according to a frequency range of the multichannel signal in order to format-convert the plurality of input channels into a plurality of output channels having elevation.

Abstract: A frequency generator for providing one or more clock signals with reduced phase jitter can include a phase-locked loop (PLL) configured to couple with a crystal and to provide a first clock signal, a multiplier circuit configured to receive the first clock signal and to provide a second clock signal, the second clock signal having a higher frequency than the first clock signal, wherein the multiplier circuit includes a second PLL, and wherein the second clock signal is an output frequency signal of the frequency generator.

Abstract: A voice processing apparatus calculates a phase difference between first and second frequency signals obtained by transforming first and second voice signals generated by two voice input units for each frequency, calculates, for each extension range set outside or inside a reference range, a presence ratio based on the number of frequencies with the phase difference between the first and second frequency signals falling within the extension range, the reference range representing a range of the phase difference between the first and second voice signals for each frequency and corresponding to a direction in which a target sound source is assumed to be located, and sets, as a non-suppression range, a first extension range having the presence ratio higher than a predetermined value and a second extension range closer to the phase difference at the center of the reference range than the first extension range is within the reference range.

Abstract: A sound device includes a housing with an accommodation space; a speaker unit located in the accommodation space and including a diaphragm for generating sound; a front cavity formed by the diaphragm and the housing; a sound channel communicated with the front cavity; and a sound-absorbing member made of sound-absorbing material located in the sound channel for covering a cross-section of the sound channel. The housing includes a bottom wall and a side wall extending from the bottom wall. The sound channel is configured to penetrate the side wall of the housing.

Abstract: The present application relates in one aspect to a method of controlling diaphragm excursion of an electrodynamic loudspeaker. The method comprises dividing the audio input signal into at least a low-frequency band signal and a high-frequency band signal by a band-splitting network and applying the low-frequency band signal to a diaphragm excursion estimator. The instantaneous diaphragm excursion is determined based on the low-frequency band signal. The determined instantaneous diaphragm excursion is compared with an excursion limit criterion. The low-frequency band signal is limited based on a result of the comparison between the instantaneous diaphragm excursion and the excursion limit criterion to produce a limited low-frequency band signal which is combined with the high-frequency band signal to produce an excursion limited audio signal.

Abstract: A noise reduction audio reproducing method includes the steps of: generating, from an audio signal of collected and obtained noise, an audio signal for noise cancellation to cancel the noise by synthesizing the audio signal for noise cancellation and the noise in an acoustic manner, reproducing the audio signal for noise cancellation acoustically to synthesize this with the noise in an acoustic manner; emphasizing an audio component to be listened to, of collected audio; synthesizing an audio signal with the audio component to be listened to being emphasized, and the audio signal for noise cancellation to supply the synthesized signal thereof to an electro-acoustic converting unit; and controlling so as to supply an audio signal, with the audio component to be listened to having been emphasized, to a synthesizing unit, regarding only a section based on a control signal.

Abstract: A hearing aid capable of detecting contact vibration noise from a collected sound signal. The hearing aid is provided with two microphones, a vibration component extracting section that extracts from collected sound signals respectively obtained by the two microphones an uncorrelated component between two collected sound signals as a vibration component for each frequency band. Additionally, a vibration noise identifying section determines whether or not a contact noise occurs based on the vibration component for each frequency band extracted by the vibration component extracting section, an acoustic signal processing section, when generating an acoustic signal by hearing aid processing of the two collected sound signals, processes the acoustic signal depending on the presence or absence of the occurrence of the contact vibration noise, and a receiver converts the acoustic signal to sound.

Abstract: Embodiments of the present invention disclose a sensitivity calibration method and an audio device. The method of the present invention includes determining whether a first signal captured in a current frame by a primary capture module is a circuit noise. When the first signal is not a circuit noise, if the first signal has a stationary noise characteristic, a first calibration gain is determined according to the first signal and a second signal captured in the current frame by a secondary capture module. The second signal is calibrated according to the first calibration gain, so that sensitivity of the primary capture module is consistent with sensitivity of the secondary capture module.

Abstract: A method and system for multiplexing audio signals into a single channel uses frequency division multiplexing. The frequency division multiplexing method herein is based on a frequency transform algorithm using FFT shifting that does not require a carrier signal for the modulation. In one embodiment, two input microphone audio signals are frequency shifted and the resulting single audio channel is directed over a standard input connection to a computing device, for instance, a smart phone using a wired TRRS connection. The TRRS analog input of the computing devices exhibits a high-pass characteristic, and the frequency shifting method enables the low frequency audio components of input audio signals to be received and processed by a software application on the smart phone. Other embodiments are disclosed.

Abstract: A diffracted sound reduction device includes: a reproduction speaker that outputs reproduced sound having properties indicated by an input signal; control speakers each of which reproduces corresponding one of control signals, the diffracted sound being a part of the reproduced sound and arriving at corresponding one of the control points except the control point at the listener's position; and control filters each of which filters the input signal to generate corresponding one of the control signals. Each of the control points faces a corresponding speaker from among the reproduction speaker and the control speakers. Each of the control filters generates the corresponding one of the control signals so that a sound pressure of the diffracted sound at corresponding one of the control points is lower than a sound pressure of direct sound that is a part of the reproduced sound which arrives at the control point of the listener's position.

Abstract: A method is provided for optimizing acoustic localization at one or more listening positions in a listening environment such as, but not limited to, a vehicle passenger compartment. The method includes generating a sound field with a group of loudspeakers assigned to at least one of the listening positions, the group of loudspeakers including first and second loudspeakers, where each loudspeaker is connected to a respective audio channel; calculating filter coefficients for a phase equalization filter; configuring a phase response for the phase equalization filter such that binaural phase difference (??mn) at the at least one of the listening positions or a mean binaural phase difference (m??mn) averaged over the listening positions is reduced in a predefined frequency range; and filtering the audio channel connected to the second loudspeaker with the phase equalization filter.

Abstract: An array loudspeaker includes a plurality of drivers arranged in an array configuration. A digital signal processor-based control system processes direct audio signal and indirect audio signal inputs for the loudspeaker to simultaneously produce direct sound in the form of a directed beam or wavefront, and indirect sound as a perceptually diffuse soundfield.

Abstract: A computer-implemented method of determining a configuration for an audio processing operation, wherein the audio processing operation comprises a predetermined set of one or more audio processing sub-operations, each audio processing sub-operation being configurable with one or more respective configuration parameters, the method comprising: specifying the predetermined set of one or more audio processing sub-operations; specifying a target frequency response; and performing a convergent optimization process to determine a configuration for the audio processing operation that reduces a difference between the frequency response of the audio processing operation and the target frequency response, wherein the configuration comprises a respective value for each configuration parameter of each audio processing sub-operation.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: An audio signal processing device receives a plurality of audio signals via a left channel (L) and a right channel (R) so as to produce a composite signal L+R and a difference signal L?R. The composite signal L+R is changed in phase with an all-pass filter, while the difference signal L?R is changed in phase and frequency characteristic with a band-pass filter (e.g. a center frequency of 1 kHz). The band-pass filter has a gently curved frequency characteristic achieving a broad passing band. Additionally, a phase difference of 90 degrees is maintained between the all-pass filter and the band-pass filter over the entire audio frequency range. The composite signal and the difference signal are adjusted in their levels and then mixed together to produce a monaural signal achieving an audio surround effect for widely propagating sound into the surrounding space without degrading sound quality.

Abstract: The sound reproduction device is applied to an acoustic space such as a passenger compartment, and controls the levels of the reproduced sounds at two evaluation points set at the seats in the passenger compartment, for example. Specifically, the sound reproduction device controls the phase of the one channel sound signal inputted from external and supplies it to the pair of speakers.

Abstract: Provided is a method and apparatus for focusing sound using an array speaker system. The method includes generating a plurality of delayed signals to be focused to a predetermined position from an input signal, filtering a low-frequency signal having a frequency that is lower than a reference frequency from the delayed signals, generating low-frequency focusing signals divided into 2 groups by adjusting a gain of the filtered low-frequency signal, and applying the low-frequency focusing signals divided into the 2 groups to speaker units of the array speaker system at both sides with respect to a center portion of the array speaker system and outputting the low-frequency focusing signals through the speaker units. In this way, the performance of sound focusing for the low-frequency signal can be improved and thus a listener located a predetermined distance from and in a predetermined direction relative to the array speaker system can clearly listen to the low-frequency focusing signals.

Abstract: The disclosed BTSC encoder includes a left high pass filter means for receiving a digital left channel audio signal and for digitally high pass filtering the digital left channel audio signal and thereby generating a digital left filtered signal; a right high pass filter means for receiving a digital right channel audio signal and for digitally high pass filtering the digital right channel audio signal and thereby generating a digital right filtered signal; a matrix means for receiving the digital left and digital right filtered signals, and including means for summing the digital left and digital right filtered signals and thereby generating a digital sum signal, and including means for subtracting one of the digital left and digital right filtered signals from the other of the digital left and digital right filtered signals and thereby generating a digital difference signal; a difference channel processing means for digitally processing the digital difference signal; and a sum channel processing means for d

Abstract: A system for actively reducing noise at a listening point, includes an earphone housing, a transmitting transducer, a receiving transducer and a controller. The transmitting transducer converts a first electric signal into a first acoustic signal, and radiates the first acoustic signal along a first acoustic path having a first transfer characteristic and along a second acoustic path having a second transfer characteristic. The receiving transducer converts the first acoustic signal and ambient noise into a second electrical signal. The controller compensates for the ambient noise by providing a noise reducing electrical signal to the transmitting transducer. The noise reducing electrical signal is derived from a filtered electrical signal that is provided by filtering the second electrical signal with a third transfer characteristic. The second and the third transfer characteristics together model the first transfer characteristic.

Abstract: Provided is a signal separation system including a rendering unit which receives a first and a second input signal and positions the first input signal according to rendering information.

Abstract: A microphone array device includes a first sound reception unit configured to obtain a first sound signal that is input from a first microphone, a second sound reception unit configured to obtain a second sound signal that is input from a second microphone, a noise state evaluation unit configured to compare the first sound signal and the second sound signal and to obtain an evaluation parameter to evaluate an influence of a non-target sound included in the second sound signal on a target sound included in the first sound signal according to a result of the comparison, a subtraction adjustment unit configured to set a suppression amount for the second sound signal based on the evaluation parameter and to generate a third sound signal; and a subtraction unit configured to generate a signal to be output based on the third sound signal and the first sound signal.

Abstract: A speaker array apparatus includes a speaker array and a control section that controls audio signals corresponding to channels respectively so that the speaker array emits sound beams for the audio signal of a part of the channels in a plurality of directions and sound beams for the audio signals of the channels other than the part of the channels in a plurality of directions. One of the sound beams for the audio signal of the part of the channels is substantially identical in direction with one of the sound beams for the audio signals of the channels other than the part of the channels. Sound images for emitting sounds for the audio signals of the channels other than the part of the channels are formed and a phantom sound image for the audio signal of the part of the channels is formed at a position which is deviated from positions of the sound images.

Abstract: A method of operating a hearing device monitors for a head movement of a hearing device wearer. When a head movement is detected, an acoustic signal acquired by a microphone is output in an amplified and phase-modulated manner with a receiver. Furthermore, a hearing device includes a motion sensor and a signal processing unit that is configured to execute the method.

Abstract: A signal processing device for supplying sound signals to a first speaker for emitting a sound on a low-pitched sound side and a second speaker for emitting a sound on a high-pitched sound side higher than the low-pitched sound side, includes a control portion that controls a characteristic of the sound signal supplied to the first speaker so that an output level of the first speaker is maintained at a crossover frequency between the first speaker and the second speaker, when an output level of the sound signal supplied to the first speaker is changed.

Abstract: A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

Abstract: The invention relates to an audio device for use proximate a user's ears. The audio device includes first and second audio transmitting/receiving devices that are capable of operating in stereo. The audio device may be used within a system for manipulating audio signals received by the device. The manipulation may include processing received audio signals to enhance their quality. The processing may include applying one or more audio enhancement algorithms such as beamforming, active noise reduction, etc. A corresponding method for manipulating audio signals is also disclosed.

Abstract: An acoustic apparatus which outputs test signals from a plurality of speakers configuring a multichannel reproduction system and adjusts acoustic characteristics by picking up response signals from the speakers using a microphone, includes a test signal storage unit storing test signals output from the speakers; a response signal storage unit storing response signals from the speakers picked up by the microphone; a parameter calculation unit calculating acoustic adjustment parameters including at least a polarity reversal parameter and a phase filter parameter based on the response signals from the speakers stored in the response signal storage unit; and an acoustic adjustment parameter storage unit storing the acoustic adjustment parameters calculated by the parameter calculation unit. The parameter calculation unit calculates the polarity reversal parameter using a low-pass component of the response signals.

Abstract: Apparatus for reproducing sound including at least three loudspeakers (5, 6, 7) mounted in a substantially sealed enclosure. The three loudspeakers may be mounted to a wall of the enclosure so that they are all directed away from, and evenly spaced around, a common point. Two speakers may be driven with the respective out of phase signals comprising the difference between two stereo channels and another speaker with the sum of those channels.

Abstract: Various embodiments relate to a system and method for decorrelating an audio signal with a hybrid filter. The hybrid filter is generated by first generating a decorrelation filter. A frequency-dependent warping is applied to the decorrelation filter. The warped decorrelation filter is then mixed with a carrier filter to generate the hybrid filter. The carrier filter may include filters for spatial processing of an audio signal, filters for upmixing an audio signal, and/or filters for downmixing an audio signal.

Abstract: A sound pickup device is provided that includes a first housing, a second housing, a first microphone, and a second microphone. The second housing is coupled to the first housing and is configured to change positions with respect to the first housing. The first microphone is mounted on the first housing and is configured to output a first audio signal based on sound picked up by the first microphone. The second microphone is mounted on the second housing and is configured to output a second audio signal based on sound picked up by the second microphone.

Abstract: In an adaptive phase discovery system a first audio signal is received via a first microphone and a second signal is received via a second microphone. Corresponding audio frames of the first and second signals are each transformed into the frequency domain and a plurality of frequency sub-bands are generated. A phase is determined for each frequency sub-band in each signal. Instantaneous phase differences are determined between the signals at each of the frequency sub-bands. Lower frequency instantaneous phase differences are filtered over time to determine current phase differences at lower frequencies. When SNR is high in lower frequency sub-bands, lower frequency sub-band phase differences are tracked to the higher frequency sub-bands. The tracked higher frequency phase differences are filtered over time to determine phase differences for the current frame. The phase differences may be used to rotate phases in each sub-band and sum signals and/or to reject off-axis signals.