Abstract: Systems and methods are described for determining orientation of an external audio device in a video conference, which may be used to provide congruent multimodal representation for a video conference. A camera of a video conferencing system may be used to detect a potential location of an external audio device within a room in which the video conferencing system is providing a video conference. Within the detected potential location, a visual pattern associated with the external audio device may be identified. Using the identified visual pattern, the video conferencing system may estimate an orientation of the external audio device, the orientation being used by the video conferencing system to provide spatial audio video congruence to a far end audience.

Abstract: It is possible to provide a noise suppression method, device, and program capable of realizing a sound image positioning of an output side corresponding to an input side with a small calculation amount. The device includes a common suppression coefficient calculation unit for receiving conversion outputs from a plurality of channels and calculating a suppression coefficient common to the channels.

Abstract: A sound source separating apparatus may include: a housing; a plurality of microphones positioned on the housing; a plurality of sound guides positioned on the housing to be adjacent to the plurality of microphones, and configured to guide sound to the plurality of microphones and to generate a difference between a plurality of sound information respectively arriving at the plurality of microphones according to a direction of a sound source; and a processor configured to separate the sound source according to the direction of the sound source based on the plurality of sound information received by the plurality of microphones.

Abstract: A method of operating a hearing apparatus and hearing apparatus having at least one of a first microphone or a second microphone which generate a first microphone signal and a second microphone signal respectively, the first microphone and the second microphone being arranged in at least one of a first hearing device and a second hearing device, a third microphone which generates a third microphone signal, the third microphone being arranged in an external device, and a signal processing unit, wherein in the signal processing unit the third microphone signal and at least one of the first microphone signal or the second microphone signal are processed together thereby producing an output signal with an enhanced signal to noise ratio compared to the first microphone signal and/or the second microphone signal.

Abstract: A device monitors, by using a microphone array, a sound generated by a monitoring target. In response to detecting a sound signal by the microphone array, the device determines a sound source direction corresponding to detected sound signal according to a sound phase difference obtained by each microphone in the microphone array. The device performs infrared detection in the sound source direction by using the infrared transceiver. The device determines a distance between the monitoring target and the infrared transceiver in the sound source direction according to an infrared detection result obtained by the infrared transceiver. The device generates location information of the monitoring target according to the sound source direction and the distance.

Abstract: In one example, a headset obtains a first audio signal including a user audio signal from a first microphone on the headset and a second audio signal including the user audio signal from a second microphone on the headset. The headset derives a first candidate signal from the first audio signal and a second candidate signal from the second audio signal. Based on the first audio signal and the second audio signal, the headset determines that a mechanical touch noise is present in one of the first audio signal and the second audio signal. In response to determining that the mechanical touch noise is present in one of the first audio signal and the second audio signal, the headset selects an output audio signal from a plurality of candidate signals including the first candidate signal and the second candidate signal. Headset provides the output audio signal to a receiver device.

Abstract: An audio signal noise estimation method includes: for multiple preset sampling points, a noise Steered Response Power (SRP) value of a Microphone (MIC) array at each preset sampling point within a preset noise sampling period is determined to obtain a noise SRP multidimensional vector including the multiple noise SRP values corresponding to the multiple preset sampling points; a present frame SRP value for a present frame of an audio signal acquired by the MIC array at each preset sampling point is determined to obtain a present frame SRP multidimensional vector including the multiple present frame SRP values corresponding to the multiple preset sampling points; and whether the audio signal acquired by the MIC array in the present frame is a noise signal is determined according to the present frame SRP multidimensional vector and the noise SRP multidimensional vector.

Abstract: A method, apparatus and computer program wherein the method comprises: obtaining a beamforming signal using respective signals from a first microphone and a second microphone; reducing the data size of the beamforming signal by grouping the beamforming signal into frequency bands and obtaining a data value for each of the frequency bands; and forming a bit stream comprising at least the reduced size beamforming signal and the signal from the first microphone wherein the bit stream enables parameters of a beamed audio channel to be controlled.

Abstract: A dual omnidirectional microphone array noise suppression is described. Compared to conventional arrays and algorithms, which seek to reduce noise by nulling out noise sources, the array of an embodiment is used to form two distinct virtual directional microphones which are configured to have very similar noise responses and very dissimilar speech responses. The only null formed is one used to remove the speech of the user from V2. The two virtual microphones may be paired with an adaptive filter algorithm and VAD algorithm to significantly reduce the noise without distorting the speech, significantly improving the SNR of the desired speech over conventional noise suppression systems.

Abstract: A vehicle and a method of controlling the vehicle are capable of outputting a sound similar to an original sound by modulating a sound signal obtained using a plurality of microphones of an external terminal device considering the physical characteristics of a user and the sound reproduction space. The vehicle includes: a speaker; a sensor module configured to detect a state of the vehicle; a communicator including a plurality of microphones and configured to receive a sound signal having two channels, information about sensitivity of each microphone of the plurality of microphones, and information about a relative position between the plurality of microphones; and a controller configured to: modulate the sound signal on at least one of the sensitivity of the plurality of microphones, the relative position between the plurality of microphones, and a Head-Related Transfer Function (HRTF); remove the reverberation from the modulated sound signal; and control the speaker.

Abstract: Provided are a sound source separation apparatus and a sound source separation method. The sound source separation apparatus includes a plurality of directional vibrators configured such that one or more of the plurality of directional vibrators react to a sound based on a direction of the sound. The sound source separation apparatus is configured to determine directions of a first sound source and a second sound source that are different from each other, based on strengths of output signals of the plurality of directional vibrators, and select a first directional vibrator and a second directional vibrator that are different from each other from among the plurality of directional vibrators to separately obtain a sound coming from the first sound source and a sound coming from the second sound source.

Abstract: An example method of operation may include designating sub-regions which collectively provide a defined reception space, receiving audio signals at a controller from the microphone arrays in the defined reception space, configuring the controller with known locations of each of the microphone arrays, assigning each of the sub-regions to at least one of the microphone arrays based on the known locations, and creating beamform tracking configurations for each of the microphone arrays based on their assigned sub-regions.

Abstract: A grille and an acoustic device having a superior exterior design without impairment of acoustic efficiency are provided. The grille includes: a through hole-provided region having a plurality of through holes; and a non-through hole-provided region having a plurality of non-through holes each with a bottom, and provided adjacent to the through hole-provided region. The through holes and the non-through holes entirely provide a plurality of openings on an exterior surface of the grille. In the non-through hole-provided region, a depth of the non-through holes on a side close to the through hole-provided region is greater than a depth of the non-through holes on a side distant from the through hole-provided region.

Abstract: A system includes an audio transducer. The audio output of the transducer may be directed at an operator. The directionality of the audio output may ensure privacy in audio delivery. Further, the directionality of the audio output may reduce the potential for other nearby individuals to be disturbed by the audio output. A directed audio system may control the content of the audio output. The content of the audio output may be configured for applications in individual operator workspaces, multiple-operator common spaces, shared-use spaces or a combination thereof. The directed audio system may customize the audio output in accord with a stored audio profile for the operator.

Abstract: This specification relates to robots and audio processing in robots. In general, one innovative aspect of the subject matter described in this specification can be embodied in a robot that includes: a body and one or more physically moveable components; a plurality of microphones and one or more other sensor subsystems; one or more processors; and one or more storage devices storing instructions that are operable, when executed by the one or more processors, to cause the robot to perform operations. The operations can include: receiving one or more sensor inputs from the one or more other sensor subsystems; determining a predicted direction of a detected sound emitter based on the one or more sensor inputs of the one or more other sensor subsystems; calculating a spatial filter based on the predicted direction; obtaining, by the plurality of microphones, respective audio inputs; and processing the respective audio inputs according to the calculated spatial filter.

Abstract: The present disclosure provides a method and system of recording an audio event which comprises a plurality of audio sources. This comprises: synchronizing a plurality of audio recording devices prior to commencement of the audio event, each recording device being suitably disposed to record audio from a single audio source and operable to provide a single audio source recording for the duration of the audio event; receiving the single audio source recording provided by each of the said audio recording devices; and processing the plurality of received single audio source recordings to synchronously combining the plurality of single audio source recordings to construct a single audio event recording.

Abstract: A method for non-invasively resolving electrophysiological activity in sub-cortical structures located deep in the brain by comparing amplitude-insensitive M/EEG field patterns arising from activity in subcortical and cortical sources under physiologically relevant sparse constraints is disclosed. The method includes a sparse inverse solution for M/EEG subcortical source modeling. Specifically, the method employs a subspace-pursuit algorithm rooted in compressive sampling theory, performs a hierarchical search for sparse subcortical and cortical sources underlying the measurement, and estimates millisecond-scale currents in these sources to explain the data. The method can be used to recover thalamic and brainstem contributions to non-invasive M/EEG data, and to enable non-invasive study of fast timescale dynamical and network phenomena involving widespread regions across the human brain.

Abstract: Systems and methods include a wind detector to receive audio input signals and output a wind detection flag including a single channel wind detection flag and a cross channel wind detection flag, each wind detection flag indicating a presence or absence of wind noise, and a fusion smoothing module to receive the plurality of wind detection flags and generate an output wind detection flag. Microphones generate the plurality of audio input signals. The wind detector and the fusion smoothing module may comprise program instructions stored in the memory for execution by a digital signal processor. The wind detector is a single channel detector to receive a single audio channel of the audio input signal and generate the single channel wind noise flag, and a cross-channel detector to compute auto correlations and a cross correlation between two or more audio channels.

Abstract: Embodiments are disclosed relating to an active noise reducing system and method for a headphone with a rigid cup-like shell which has an outer surface and an inner surface that encompasses a cavity with an opening. The system and method include picking up sound at least at three positions that are regularly distributed over the outer surface, and providing a first electrical signal that represents the picked-up sound. The system and method further include: filtering the first electrical signal to provide a second electrical signal, and generating in the opening of the cavity sound from the second electrical signal. Filtering is performed with a transfer characteristic that is configured so that noise that travels through the shell from beyond the outer surface to beyond the inner surface is reduced by the sound generated in the opening.

Abstract: A sensor information acquiring device includes a sensor and a control section configured to determine that a position of a target object does not change even if a relative relation between the sensor and the target object changes and control an information acquisition direction of the sensor on the basis of information concerning displacement of a displacement section to which the sensor is attached. Even when a position, a direction, and the like of the sensor are changed by influence of work and the like, the sensor information acquiring device can always acquire effective sensor information concerning a desired target object.

Abstract: A device, system, and method for modeling microphones and a microphone sound-isolation baffle that can be used with microphone modeling. The microphone modeling device, system, and method, can account for the effects of a microphone modeled with a microphone sound-isolation baffle and reduce unwanted audio coloration. The microphone model can work with single-capsule and dual-capsule microphones with the dual-capsule modeling able to achieve greater off-axis rejection and reduced off-axis coloration. The microphone modeling microphone sound-isolation baffle can attach to a specific reference microphone used for microphone modeling. The microphone sound-isolation baffle can be designed so the filter only attaches at a predetermined distance and at a predetermined rotational angle with respect to the microphone.

Abstract: A user terminal device includes a touch screen including a main display area and an auxiliary display area including a curved portion extending from the main display area, and a processor configured to, in response to an input for moving an icon displayed on the main display area to the auxiliary display area, control the touch screen to display the icon on the auxiliary display area.

Abstract: Methods and apparatus to improve the accuracy of crediting media exposure through detecting reverberation indicative of spillover are disclosed. An example apparatus includes a reverberation analyzer to identify a quantity of short durations of loudness in an audio signal of media presented by a media presentation device and calculate a ratio of the quantity of the short durations of loudness to a quantity of durations of loudness in the audio signal of the media, the quantity of the durations of loudness including the quantity of short durations of loudness. The example apparatus also includes a processor and memory in circuit with the processor, the memory including instructions that, when executed by the processor, cause the processor to mark the media as un-usable to credit a media exposure when the ratio does not satisfy a loudness ratio threshold.

Abstract: A detection target portion of a device is captured by an image capturing unit. An operation unit of the device is operated by a user and causes the detection target portion to move in conjunction with the operation unit in accordance with the operation. A detection target portion image representing the detection target portion comprised in a captured image captured by the image capturing unit is detected, and at least one of a position and a shape of the detection target portion image in the captured image is specified. Then, based on the at least one of the position and the shape of the detection target portion image, it is determined whether or not a condition for performing a first process that is information processing is satisfied. Then, based on data output from an inertial sensor, the condition is changed.

Abstract: A method improves speech recognition using a device located in proximity to a machine emitting high levels of audio noise. The microphone of the device receives the audio noise emitted by the machine and the speech emitted by a user and generates a composite signal. The device also receives a wireless communication signal from the machine comprising information on an audio noise profile and the proximity of the machine relative to the device. The audio noise profile is a representation of the audio noise emitted by the machine. Based on this information, the device determines a filter for filtering the composite signal to mitigate the audio noise before initiating the speech recognition process. The method improves speech recognition in a high audio noise environment.

Abstract: The disclosure present a method for determining a status of one or more features of a hearing aid device and a hearing system, wherein the hearing system includes the hearing aid device, and the hearing aid device comprises a housing part, a connection part and an ear piece connected via the connection part to the housing part, and wherein the hearing aid device comprising; a first microphone configured to receive a first acoustic signal and provide a first audio signal based on the first acoustic signal, a signal processor connected to the first microphone and configured to receive the first audio signal and provide an output audio signal based on the first audio signal, a speaker configured to receive the output audio signal and output an acoustic output signal via the ear piece, an anti-feedback unit configured to receive the output audio signal from the signal processor and a secondary first audio signal from the first microphone, and the anti-feedback unit is further configured to estimate a first feedba

Abstract: There is provided a clip-on microphone assembly comprising a clip-on mechanism (16) for attaching the microphone assembly (10) to the cloths of a user (11); at least three microphones (20, 21, 22) for capturing audio signals from the user's voice, the microphones defining a microphone plane; an acceleration sensor (32) for generating an orientation signal by sensing gravitational acceleration in at least two orthogonal dimensions, an audio signal processing unit (34) for producing an output audio signal (36) from the captured audio signals, comprising a beamformer unit (24) for processing the captured audio signals, in order to generate the output audio signal, in a manner so as to create an acoustic beam having a direction, a DOA unit (30) for determining the direction of arrival of sound by analyzing the captured audio signals; and a control unit (26) for controlling the beamforming unit, the control unit being adapted to determine an allowed angular sector (40) of the direction of the acoustic beam accordi

Abstract: Systems and methods for utilizing microphone array information for acoustic modeling are disclosed. Audio data may be received from a device having a microphone array configuration. Microphone configuration data may also be received that indicates the configuration of the microphone array. The microphone configuration data may be utilized as an input vector to an acoustic model, along with the audio data, to generate phoneme data. Additionally, the microphone configuration data may be utilized to train and/or generate acoustic models, select an acoustic model to perform speech recognition with, and/or to improve trigger sound detection.

Abstract: A method comprising: receiving a first microphone signal from a first microphone having a first frequency response characteristic (1101, 1121) at frequencies (114) associated with wind noise; receiving a second microphone signal from a second microphone having a second frequency response characteristic (1102, 1122) at frequencies (114) associated with wind noise, wherein the first frequency response characteristic (1101, 1121) provides less gain than the second frequency response characteristic (1102, 1122) over the range of frequencies (114) associated with wind noise; and processing the first microphone signal and the second microphone signal to detect the presence of wind noise.

Abstract: A digital signal processor is programmed to extract a center channel from a stereo input, apply the center channel to an array of speaker elements using a first set of finite impulse response filters and a first rotation matrix to generate a first beam of audio content at a target angle about the axis, apply a left channel of the stereo input to the array of speaker elements using a second set of finite impulse response filters and a second rotation matrix to generate a second beam of audio content at a first offset angle from the target angle about the axis, and apply a right channel of the stereo input to the array of speaker elements using a third set of finite impulse response filters and third rotation matrix to generate a third beam of audio content at a second offset angle from the target angle about the axis.

Abstract: The present disclosure provides a robot and an audio data processing method thereof. The robot includes a body part, a main control module, and a sound pickup module electrically coupled to the main control module. The sound pickup module includes N microphones distributed around the body part to collect audio data. The main control module is configured to obtain the audio data of a sound source from a part of the N microphones collecting the audio data of the sound source without blocked by the body part, and perform a sound source localization and a sound pickup based on the obtained audio data. The 360-degree wake-up and sound source localization of the robot and the beam-forming of directional beams are realized. In addition, the sound pickup is realized without forming microphone holes on the head of the robot, hence the aesthetics of the robot will not be affected.

Abstract: A method for operating a hearing device. In the hearing device a first directional signal and a second directional signal are generated from an ambient sound signal. The first directional signal and the second directional signal are used to determine at a first response time a first adaptation coefficient for a first superposition of the first directional signal with the second directional signal for the purpose of noise suppression. It is intended here that the first directional signal and the second directional signal are used to determine at a second response time a second adaptation coefficient for a second superposition of the first directional signal with the second directional signal for the purpose of noise suppression. The first adaptation coefficient and the second adaptation coefficient are used to determine an output adaptation coefficient for forming an output signal by superposition of the first directional signal and the second directional signal.

Abstract: An acoustic spatial diagnostic circuit couples to an array of microphones to successively sample an acoustic environment surrounding a wireless transceiver to generate an acoustic spatial map of activity within the environment based on sets of acoustic samples and execute one or more actions based on a related portion of the acoustic spatial map exhibiting one or more correlations with a spatial context condition of a wireless trained transceiver.

Abstract: An apparatus comprising at least one processor and at least one memory including computer program code the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to perform processing at least one control parameter dependent on at least one sensor input parameter, processing at least one audio signal dependent on the processed at least one control parameter, and outputting the processed at least one audio signal.

Abstract: The present technology relates to a sound source separation apparatus and a method which make it possible to separate a sound source at lower calculation cost. A communication unit receives a spatial frequency spectrum of a sound collection signal which is obtained by a microphone array collecting a plane wave of sound from a sound source, and a spatial frequency mask generating unit generates a spatial frequency mask for masking a component of a predetermined region in a spatial frequency domain on the basis of the spatial frequency spectrum. A sound source separating unit extracts a component of a desired sound source from the spatial frequency spectrum as an estimated sound source spectrum on the basis of the spatial frequency mask. The present technology can be applied to a spatial frequency sound source separator.

Abstract: A system performs an optimization algorithm to optimize two or more acoustic sensors of a microphone array. The system obtains an array transfer function (ATF) for a plurality of combinations of the acoustic sensors of the microphone array. In a first embodiment, the algorithm optimizes an active set of acoustic sensors on the eyewear device. The plurality of combinations may be all possible combinations of subsets of the acoustic sensors that may be active. In a second embodiment, the algorithm optimizes a placement of two or more acoustic sensors on an eyewear device during manufacturing of the eyewear device. Each combination of acoustic sensors may represent a different arrangement of the acoustic sensors in the microphone array. In each embodiment, the system evaluates the obtained ATFs and, based on the evaluation, selects a combination of acoustic sensors for the microphone array.

Abstract: A beamforming audio capture apparatus comprises a microphone array (301) which is coupled to a first beamformer (303) and a second beamformer (305). The beamformers (303, 305) are filter-and-combine beamformers comprising a plurality of beamform filters each having an adaptive impulse response. A difference processor (309) determines a difference measure between beams of the first beamformer (303) and the second beamformer (305) in response to a comparison of the adaptive impulse responses of the two beamformers (303, 305). The difference measure may e.g. be used to combine the output signals of the beamformers (303, 305). An improved difference measure less sensitive to e.g. diffuse noise may be provided.

Abstract: An electronic device may include a housing and a directional microphone. The housing may include a front surface, a rear surface positioned opposite the front surface, and a hollow cavity positioned between the front surface and the rear surface. The hollow cavity may include a front opening defined in the front surface and a rear opening defined in the rear surface. The directional microphone may include a front port and a rear port. The directional microphone may be mounted in the hollow cavity of the housing with the front port oriented toward the front opening and with the rear port oriented toward the rear opening.

Abstract: Techniques for improving acoustic echo cancellation to attenuate an echo signal generated by a wireless loudspeaker are described. A relative position of the wireless loudspeaker is determined when sound from the wireless loudspeaker is determined to be the dominant sound (e.g., during output of a response to a user command or query or during speech by a far-side party during a two-way audio communication). A beam corresponding to the relative position is thereby selected for echo cancellation.

Abstract: Input audio data portions of a common time window index value generated by multiple microphones at a location are received. Subband portions are generated from the input audio data portions. Peak powers, noise floors, etc., are individually determined for the subband portions. Weights for the subband portions are computed based on the peak powers, the noise floors, etc., for the subband portions. An integrated audio data portion of the common time window index is generated based on the subband portions and the weight values for the subband portions. An integrated signal may be generated based at least in part on the integrated audio data portion.

Abstract: The disclosure relates to a microphone system for a motor vehicle, comprising a microphone housing in which a first microphone, a second microphone and a third microphone are disposed, with a signal processing device, which is designed to process respective signals provided by the microphones, wherein the signal processing device is configured to receive the signals of the first and the second microphones in a first operating mode in such a way that a driver directivity, which is oriented towards a first position of a driver's seat of the motor vehicle is provided, and the signals of the second and third microphones in that a passenger directivity, which is aligned with a first position of a passenger's seat of the motor vehicle is provided in the first operating mode, wherein the signal processing device is configured to process the signals of the first and the third microphones in a second operating mode in such a way that an alternative driver directivity is provided, which is oriented towards a second pos

Abstract: An earpiece for use in voice dictation includes a speaker disposed within the earpiece housing, a microphone, and a processor disposed within the earpiece housing and operatively connected to the microphone and the speaker, wherein the processor is adapted to capture a voice stream from the microphone. The earpiece may further include a wireless transceiver disposed within the earpiece housing, the wireless transceiver operatively connected to the processor. The earpiece is configured to be controlled by a user through a plurality of different user interfaces to perform voice dictation.

Abstract: Microphone array calibration that does not require a calibrated source or calibrated reference microphone is provided. We provide a statistical (Bayesian) algorithm that (under condition of reasonable environment noise during calibration) can determine gain and phase differences of a whole array at once, even when the gain and/or phase of the source is unknown. More specifically, a Bayesian regression with complex log-normal prior and complex normal likelihood is employed. The inherent phase-wrapping ambiguity in this regression is resolved by exploiting the similarity of likelihood between a lattice point and its Euclidean Voronoi region.

Abstract: Provided is a sound source localization method including steps of: (a) receiving a mixed signal of a target sound source signal and noise and echo signals through multiple microphones including at least two microphones; (b) generating a binarized mask based on a diffuseness by using a coherence-to-diffuseness ratio CDR, which is information on the target sound source and the noise source, by using the input signal; (c) pre-processing an input signal to multiple microphones by using the generated binarized mask; and (d) performing a predetermined algorithm such as the GCC-PHAT or the SRP-PHAT on the pre-processed input signal to estimate a direction of the target sound source.

Abstract: Systems and methods are disclosed herein for modifying modulated signals for transmission. The system receives a modulated signal comprising a speech signal and a carrier wave and generates first and second spectral signals by converting the modulation signal and carrier wave from the time domain to the frequency domain respectively. The system then determines spectral bands for the first and second spectral signals. For each spectral band, the system calculates a weighted spectral band value based on a magnitude of the first spectral signal within the spectral band and generates a modified spectral signal by modifying the second spectral signal with the weighted spectral band value. The system then converts the modified spectral signal from the frequency domain to the time domain and transmits the converted modified spectral signal to a server.

Abstract: An integrated circuit, a circuit assembly and a method for operation the integrated circuit are disclosed. In embodiments the integrated circuit includes at least one supply voltage terminal configured to receive a supply voltage for operation the integrated circuit, at least one input terminal configured to receive an analog input signal corresponding to an audio signal, at least one output terminal configured to provide an analog output signal and a signal strength detector configured to detect a signal strength of the analog input signal provided at the at least one input terminal, wherein the integrated circuit is configured to amplify an audio signal based on the detected signal strength and to output a corresponding amplified signal at the at least one output terminal, wherein the integrated circuit comprises a signaling circuit configured to indicate an amplification setting of the integrated circuit at the at least one supply voltage terminal.

Abstract: A metamaterial comprising, a plurality of acoustic vector field sensors, each configured to sense an acoustic vector field of a fluid within a fluid-filled space in response to fluid waves, and producing an electrical signal corresponding to the sensed acoustic vector field; a processor configured to perform a time and space transform on the electrical signal; and at least one phased array transducer, configured to emit fluid waves according to a produced acoustic vector field pattern dependent on a result of the time and space transform, a within a portion of the fluid.

Abstract: The disclosed computer-implemented method may include (1) establishing a communication channel to indirectly convey a conversation, (2) receiving, via the communication channel, a portion of the conversation, (3) presenting the portion of the conversation to a user, (4) receiving, via the communication channel, an additional portion of the conversation, (5) detecting an additional communication channel capable of conveying the conversation, (6) determining a human-perceivable difference between how the conversation has been conveyed via the communication channel and how the conversation will be conveyed via the additional communication channel, and (7) compensating for the human-perceivable difference when presenting the additional portion of the conversation to the user in order to smoothly transition the conversation from the communication channel to the additional communication channel. Various other methods, systems, and computer-readable media are also disclosed.

Abstract: A system and method is described for determining whether a loudspeaker device has relocated, tilted, rotated, or changed environment such that one or more parameters for driving the loudspeaker may be modified and/or a complete reconfiguration of the loudspeaker system may be performed. In one embodiment, the system may include a set of sensors. The sensors provide readings that are analyzed to determine 1) whether the loudspeaker has moved since a previous analysis and/or 2) a distance of movement and/or a degree change in orientation of the loudspeaker since the previous analysis. Upon determining the level of movement is below a threshold value, the system adjusts previous parameters used to drive one or more of the loudspeakers. By adjusting previous parameters instead of performing a complete recalibration, the system provides a more efficient technique for ensuring that the loudspeakers continue to produce accurate sound for the listener.

Abstract: Methods and systems are provided for determining a maximum expected response of a reverberant wavefield system to an excitation, where there is uncertainty in both the excitation and the dynamic properties of the system. An exemplary method of characterizing a reverberant response associated with an reverberant subsystem involves determining a first variance associated with an excitation energy exposed to the reverberant subsystem, determining a second variance associated with an effective damping loss factor of the reverberant subsystem, determining a third variance associated with an input modal power acceptance of the reverberant subsystem—and for multiple connected subsystems, determining a fourth variance associated with a coupling loss factor of the coupled subsystems—determining a cumulative variance associated with the reverberant response based on the first variance, the second variance, and the third variance, and displaying an output influenced by the cumulative variance on a display device.