Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: Example embodiments disclosed herein relate to orientation-aware surround sound playback. A method for processing audio on an electronic device that includes a plurality of loudspeakers is disclosed, the loudspeakers arranged in more than one dimension of the electronic device. The method includes, responsive to receipt of a plurality of received audio streams, generating a rendering component associated with the plurality of received audio streams, determining an orientation dependent component of the rendering component, processing the rendering component by updating the orientation dependent component according to an orientation of the loudspeakers and dispatching the received audio streams to the plurality of loudspeakers for playback based on the processed rendering component. Corresponding system and computer program products are also disclosed.

Abstract: A method, apparatus and computer program, the method comprising; using a first microphone and a second microphone to detect ambient noise where the first microphone is positioned at a first position within a headset and the second microphone is positioned at a second position within the headset; comparing the ambient noise detected by the first microphone to the ambient noise detected by the second microphone to determine locations of the microphones; and using the determined locations of the microphones to enable a spatial audio output signal to be rendered by the heads.

Abstract: A processor and memory comprising instructions stored therein executable by the processor to display a first media item available for playback, wherein the first media item is associated at least with a preview option, a playback option, and a queue option. A first input associated with the first media item is received. The first input is determined to correspond only to the preview option. Responsive to determining that the first input corresponds only to the preview option, causing at least a portion of the first media item to be played.

Abstract: An apparatus for generating an audio output signal is provided. The audio output signal has two or more audio output channels and is generated from an audio input signal having two or more audio input channels. The apparatus includes a provider and a signal processor. The provider is adapted to provide first covariance properties of the audio input signal. The signal processor is adapted to generate the audio output signal by applying a mixing rule on at least two of the two or more audio input channels. The signal processor is configured to determine the mixing rule based on the first covariance properties of the audio input signal and based on second covariance properties of the audio output signal, the second covariance properties being different from the first covariance properties.

Abstract: A method includes: receiving a representation of a soundfield, the representation characterizing the soundfield around a point in space; decomposing the received representation into independent signals; and encoding the independent signals, wherein a quantization noise for any of the independent signals has a common spatial profile with the independent signal.

Abstract: A single enclosure multi-channel loudspeaker product 100 uses a novel signal processing system and method to achieve a surprisingly effective psycho-acoustically expanded image breadth by inter-aural crosstalk cancellation, in a manner which relies on a new method for cancellation of apparent sources of inter-aural crosstalk. In the commonly owned Polk® SDA™ (prior art) method, the optimal distance between stereo pair main and effect (SDA) loudspeakers was required to be substantially equal to the ear-to-ear width of a typical user's head. Compact SDA speaker system 100 employs digital signal processing generating selected time delays to acoustically simulate the optimal placement of an effects transducer relative to its main transducer for a physically compact configuration having each side's “main” transducer (e.g., 108LMS) spaced at less than 5.5 inches from the side's corresponding SDA (or effects) transducer (e.g., 108LSS), and this permits the system enclosure to be surprisingly compact, (e.g.

Abstract: A device includes at least one microphone to receive audio; a processor to process a signal representation of the audio to recognize speech and reduce acoustic echoes detected in the signal representation; at least one speaker mounted in the housing and oriented to output sound away from the microphone; a gateway coupled to the processor to communicate with home appliances, the gateway communicating with a plurality of home area network (HAN) protocols including Zigbee (IEEE 802.15.4), Bluetooth (IEEE 802.15.1) and Z-Wave, the gateway translating HAN data from one protocol to another and providing interoperability among appliances; and one or more home appliances having a security level requiring additional check prior to granting access by requiring video, speaker voice, or mobile device authentication.

Abstract: An apparatus and a method for controlling sound in a vehicle may include a sound detector that detects a sound output position at which the sound is output in the vehicle, and a processor that changes a sound position in the vehicle based on the sound output position detected by the sound detector.

Abstract: A method for processing an audio signal is performed at a computing device. The method includes the following steps: receiving a digital stereo audio input signal; extracting localization cues from the digital stereo audio input signal; generating a left-side component and a right-side component from the digital stereo audio input signal, at least partially, in accordance with the localization cues; performing crosstalk cancellation to the left-side component and the right-side component, respectively, to obtain a crosstalk-cancelled left-side component and a crosstalk-cancelled right-side component; and generating a digital stereo audio output signal including the crosstalk-cancelled left-side component and the crosstalk-cancelled right-side component.

Abstract: A method for processing an audio signal is performed at a computing device. The method includes the following steps: receiving a digital stereo audio input signal; extracting localization cues from the digital stereo audio input signal; generating a left-side component and a right-side component from the digital stereo audio input signal, at least partially, in accordance with the localization cues; performing crosstalk cancellation to the left-side component and the right-side component, respectively, to obtain a crosstalk-cancelled left-side component and a crosstalk-cancelled right-side component; and generating a digital stereo audio output signal including the crosstalk-cancelled left-side component and the crosstalk-cancelled right-side component.

Abstract: When a channel signal, such as a 22.2 channel signal, is rendered into a 5.1 channel signal, a three-dimensional (3D) audio may be reproduced by using a two-dimensional (2D) output channel, however, when an elevation angle of an input channel is different from a standard elevation angle, if elevation rendering parameters according to the standard elevation angle are used, distortion may occur in a sound image.

Abstract: A method for processing an audio signal is performed at a computing device. The method includes the following steps: receiving a digital stereo audio input signal; extracting localization cues from the digital stereo audio input signal; generating a left-side component and a right-side component from the digital stereo audio input signal, at least partially, in accordance with the localization cues; performing crosstalk cancellation to the left-side component and the right-side component, respectively, to obtain a crosstalk-cancelled left-side component and a crosstalk-cancelled right-side component; and generating a digital stereo audio output signal including the crosstalk-cancelled left-side component and the crosstalk-cancelled right-side component.

Abstract: A method for updating a factor graph (10;10?;10?) of an a posteriori probability estimator, the factor graph including at least one repetition node (13;13?;13?) and at least one sum node (11;11?;11?), wherein at least two connections are associated with each node, and wherein each connection is associated with an incoming message at the node and with an outgoing message from the node, wherein the method includes the steps of: storing the nodes' incoming and outgoing messages into memory (12;12?;12?) of the estimator as messages belonging to one same class of wrapped and/or sampled Gaussian messages; updating the node of the factor graph (10;10?;10?) by using a resulting message belonging to the class of incoming messages, the resulting message being obtained by processing the incoming wrapped and/or sampled Gaussian messages.

Abstract: A hearing aid fitting system (400) adapted for providing sound samples illustrating the impact on sound quality from a hearing aid system defect and a method of providing such sound samples. The invention also relates to a hearing aid system and computer program code capable of carrying out such a method of providing sound samples illustrating the impact on sound quality from a hearing aid system defect and methods of operating and fitting hearing aid systems.

Abstract: There is provided a method and corresponding system for determining phase adjustment filters for an associated sound generating system including at least two audio reproduction channels C1 and C2 where each of the audio reproduction channels C1 and C2 has an input signal and at least one loudspeaker located in a listening environment. The method includes estimating, for each of the audio reproduction channels C1 and C2, an acoustic transfer function at each of M?1 spatial positions in the listening environment, based on sound measurements at the spatial positions; and determining, based on the acoustic transfer functions, phase adjustment filters F1 and to be applied, respectively, to the audio reproduction channels C1 and C2, to reduce the inter-loudspeaker differential phase, IDP, between the audio reproduction channels C1 and C2 in p listener positions.

Abstract: The derivation of personalized HRTFs for a human subject based on the anthropometric feature parameters of the human subject involves obtaining multiple anthropometric feature parameters and multiple HRTFs of multiple training subjects. Subsequently, multiple anthropometric feature parameters of a human subject are acquired. A representation of the statistical relationship between the plurality of anthropometric feature parameters of the human subject and a subset of the multiple anthropometric feature parameters belonging to the plurality of training subjects is determined. The representation of the statistical relationship is then applied to the multiple HRTFs of the plurality of training subjects to obtain a set of personalized HRTFs for the human subject.

Abstract: Methods, systems, and apparatuses are described for determining relative locations of wireless loudspeakers and performing channel mapping thereof. An audio processing component utilizes sounds produced by wireless loudspeakers during setup/installation procedures, which are received by a microphone at locations in an acoustic space, to determine an amount of time between when the audio signal is initially transmitted and when the microphone signal is received. The audio processing component also utilizes wireless timing signals provided by a wireless transceiver, at locations in the acoustic space, to wireless loudspeakers and then back to the wireless transceiver to determine an amount of time between transmission and reception by the wireless transceiver. The timing delays are used to determine the locations of the wireless loudspeakers in the acoustic space.

Abstract: Example embodiments disclosed herein relate to audio object clustering with single channel quality preservation. A method of clustering audio objects is disclosed. The method includes determining cluster positions based on object positions of the audio objects and a reference speaker layout, the reference speaker layout indicating speakers located at different speaker positions. The method also includes determining object-to-cluster gains based on the determined cluster positions, the object positions and the reference speaker layout, an object-to-cluster gain defining a proportion of the respective audio object that is assigned to a cluster associated with one of the determined cluster positions. The method further includes clustering the audio objects based on the object-to-cluster gains and the cluster positions for generating cluster signals. Corresponding system, computer program product and device for clustering audio objects are also disclosed.

Abstract: The speaker device includes a cylindrical main body, a sound receiving portion, plural speaker units and at least one passive radiation plate. The cylindrical main body includes a top portion and a bottom portion. The sound receiving portion disposed on the top portion of the cylindrical main body. The speaker units are disposed within the cylindrical main body. The exports of the speaker units face the directions extending radially outward from an axial center of the cylindrical main body respectively. Each of the speaker units outputs a voice signal. The passive radiation plate is disposed on a surrounding surface of the cylindrical main body, and located between the speaker units and the bottom portion of the cylindrical main body. The passive radiation plate is driven to produce resonance with the air within the cylindrical main body, which is vibrated by the vibration of the voice signal.

Abstract: A system and method for contextual tagging of data on at least one display associated with a vehicle is provided. The method includes: receiving data regarding the vehicle from at least one of a source of verbal data and a source of non-verbal data; determining a context associated with the data and a variable parameter associated with the context; determining a display element associated with the at least one display that corresponds to the determined variable parameter; generating a symbol for the display to display adjacent to the determined display element based on at least one of the determined context and the determined variable parameter; generating a selectable icon for the display to render adjacent to the symbol based on received verbal data; associating the received verbal data to the selectable icon; and outputting the received verbal data to an audio device based on a selection of the selectable icon.

Abstract: Systems and methods for HRTF personalization are provided. More specifically, the systems and methods provide HRTF personalization utilizing non-parametric processing of three-dimensional head scans. Accordingly, the systems and methods for HRTF personalization generate a personalized set of HRTFs for a user without having to extract specific geometric and/or anthropometric features from a three dimensional head scan of a user and/or from the three dimensional head scans of training subjects in a database.

Abstract: Systems and methods are disclosed in which a playback device transmits a first sound signal including a predetermined waveform. In one example, the playback device receives a second sound signal including at least one reflection of the first sound signal. The second sound signal is processed to determine a location of a person relative to the playback device, and a characteristic of audio reproduction by the playback device is selected, based on the determined location of the person.

Abstract: A method convolves a voice in a telephone call as binaural sound to an external sound localization point (SLP) that is familiar to a user. The method analyzes SLPs in prior telephone calls where voices in the prior telephone calls externally localized as binaural sound to the user and determines, based on the analysis, a SLP that is familiar to the user where the user previously localized the voices in the prior telephone calls.

Abstract: An out-of-head localization processing apparatus according to an embodiment includes headphones, left and right microphones, a measurement unit configured to measure left and right headphone transfer characteristics, respectively, an inverse-filter calculation unit configured to calculate inverse filters of the headphone transfer characteristics, a correction unit configured to calculate correction filters by correcting the inverse filters, and an input unit configured to receive a user input. The correction unit corrects the inverse filters by using a predefined correction function in a first frequency band. The correction unit corrects the inverse filters according to a correction pattern selected based on the user input in a second frequency band.

Abstract: Embodiments provide a digital processor including an ambient portion extractor and a spatial effect processing stage. The ambient portion extractor is configured to extract an ambient portion from a multi-channel signal. The spatial effect processing stage is configured to generate a spatial effect signal based on the ambient portion of the multi-channel signal. The digital processor is configured to combine the multi-channel signal or a processed version thereof with the spatial effect signal.

Abstract: In at least one embodiment, an apparatus for enhancing speech prompts in a vehicle is provided. The apparatus includes an audio processor that is electrically coupled to a plurality of loudspeakers in a vehicle. The audio processor being programmed to provide entertainment data to the plurality of loudspeakers for playback in the vehicle and to receive a speech prompt indicative of only a spoken audio output to a driver in the vehicle. The audio processor is further programmed to mute the entertainment data that is played back on a first loudspeaker of the plurality of loudspeakers in response to the speech prompt and to provide the speech prompt to the first loudspeaker for playback to the driver in response to the speech prompt.

Abstract: Method for processing audio objects by an audio client apparatus is described wherein the method comprises: receiving or determining spatial listener information, the spatial listener information defining including one or more listener positions, orientations and/or foci of one or more listeners in the audio space; the audio client apparatus selecting one or more audio object identifiers from a set of audio object identifiers defined in a manifest file stored in a memory of the audio client apparatus, an audio object identifier defining an audio object being associated with audio object position information for defining one or more positions of the audio object in the audio space; the selecting of the one or more audio object identifiers by said audio client apparatus being based on the spatial listener information and the audio object position information of audio object identifiers in said manifest file; and, the audio client apparatus using said one or more selected audio object identifiers to request tran

Abstract: The methods described herein are configured to accurately propagate sound through a virtual environment. During runtime of an application using the virtual environment, a shortest path is calculated from a sound source in a first zone to a sound destination in a second zone that passes through at least one portal. The direction of the calculated shortest path from the portal to the sound destination is determined. Then, a virtual sound source is generated based on the length and determined direction of the calculated path. Further, obstructions in the virtual environment are processed to determine an attenuation of audio data associated with the sound source. The generated virtual sound source and the attenuated audio data are provided to an audio engine for rendering to a user. Propagating sound using a combination of these methods provides an accurate, realistic sound experience in the virtual environment.

Abstract: The derivation of personalized HRTFs for a human subject based on the anthropometric feature parameters of the human subject involves obtaining multiple anthropometric feature parameters and multiple HRTFs of multiple training subjects. Subsequently, multiple anthropometric feature parameters of a human subject are acquired. A representation of the statistical relationship between the plurality of anthropometric feature parameters of the human subject and a subset of the multiple anthropometric feature parameters belonging to the plurality of training subjects is determined. The representation of the statistical relationship is then applied to the multiple HRTFs of the plurality of training subjects to obtain a set of personalized HRTFs for the human subject.

Abstract: A method processes binaural sound to externally localize to a first user at a first location. This location is shared such that an electronic device processes the binaural sound to externally localize to a second user at a second location. The first and second locations occur at a same or similar location such that the first and second users hear the binaural sound as originating from the same or similar location.

Abstract: A computing system comprises a memory including instructions and a processor coupled to the memory. The processor, when executing the instructions, is configured to perform the steps of mapping, to a robotic speaker, a sound associated with a virtual object included in a virtual environment, transmitting information to the robotic speaker to cause the robotic speaker to move to a location in a physical environment that corresponds to a location of the virtual object in the virtual environment, and transmitting an audio signal to the robotic speaker. The robotic speaker outputs, based on the audio signal and at the location in the physical environment, the sound associated with the virtual object.

Abstract: Methods and apparatus assist listeners in distinguishing between electronically generated binaural sound and physical environment sound while the listener wears a wearable electronic device that provides the binaural sound to the listener. The wearable electronic device generates a visual alert or audio alert when the electronically generated binaural sound occurs.

Abstract: An apparatus includes a receiver and a decoder. The receiver is configured to receive a bitstream that includes an encoded mid channel and a quantized value representing a shift between a reference channel associated with an encoder and a target channel associated with the encoder. The quantized value is based on a value of the shift. The value of the shift is associated with the encoder and has a greater precision than the quantized value. The decoder is configured to decode the encoded mid channel to generate a decoded mid channel and to generate a first channel based on the decoded mid channel. The decoder is further configured to generate a second channel based on the decoded mid channel and the quantized value. The first channel corresponds to the reference channel and the second channel corresponds to the target channel.

Abstract: Embodiments of the invention relate generally to electrical and electronic hardware, computer software, wired and wireless network communications, and wearable computing and audio devices for communication audio. More specifically, disclosed are an apparatus and a method for processing audio signals to include spatially modulated message audio signals as a portion of a monaural signal. In some embodiments, a method includes receiving a message for a loudspeaker. The method can determine whether an audio signal is in communication with the loudspeaker and a type of a message of the message. Message audio for the message can be spatially modulated as a function of the type of message. A mono-spatial audio signal can be formed based on the audio signal and the spatially-modulated message. Thus, a monaural audio signal can be modulated to generate mono-spatial effects for presenting the messages.

Abstract: An audio cross-connect headphone with an audio cross-connect. A first embodiment has an acoustical cross-connect duct. A second embodiment has a first time delay circuit configured generate a time delayed right stereo signal and mix the time delayed right stereo signal with a left stereo signal, and a second time delay circuit configured to generate a time delayed left stereo signal and mix the time delayed left stereo signal with the right stereo signal. A third embodiment has a Resister-Inductor-Capacitor (RLC) circuit, wherein the right speaker ground wire and left speaker ground wire both split and terminate on a first node of the RLC circuit, then both resume from the second node of the RLC circuit towards the left speaker and right speaker. A fourth embodiment, similar to the third, but the speaker ground wires from the speakers join, but do not connect to the RLC circuit.

Abstract: Systems and method discussed herein involve applying a designation of a default playback device in a media playback system. One method may involve maintaining a database comprising data indicating an assignment of an identification of a network microphone device to a playback zone of a media playback system, receiving a message indicating (i) the identification of the network microphone device and (ii) a media playback command, identifying the playback zone in the media playback system based on the data in the database and the received message, and transmitting to the playback zone, a message indicating the media playback command.

Abstract: The present technology relates to a sound field collecting apparatus and method, a sound field reproducing apparatus and method and a program which enable a sound field to be reproduced accurately at lower cost. Each linear microphone array outputs a sound collection signal obtained by collecting a sound field. A spatial frequency analysis unit performs spatial frequency transform on each sound collection signal to calculate spatial frequency spectra. A space shift unit performs space shift on the spatial frequency spectra so that central coordinates of the linear microphone arrays become the same, to obtain spatially shifted spectra. A space domain signal mixing unit mixes a plurality of spatially shifted spectra to obtain a single microphone mixed signal. By mixing the sound collection signals of the plurality of linear microphone arrays in this manner, it is possible to reproduce a sound field accurately at low cost. The present technology can be applied to a sound field reproducer.

Abstract: An apparatus for generating loudspeaker signals includes an object metadata processor configured to receive metadata, to calculate a second position of the audio object depending on the first position of the audio object and on a size of a screen if the audio object is indicated in the metadata as being screen-related, to feed the first position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being not screen-related, and to feed the second position of the audio object as the position information into the object renderer if the audio object is indicated in the metadata as being screen-related. The apparatus further includes an object renderer configured to receive an audio object and to generate the loudspeaker signals depending on the audio object and on position information.

Abstract: A synchronous audio playback method, apparatus and system are provided. The method includes calculating a first playback delay generated when an audio file is played; acquiring a second playback delay; calculating a delay offset; calculating a data adjustment amount; adding to-be-adjusted data to or deleting to-be-adjusted data from a current to-be-played part of the audio file according to a value of the delay offset, to obtain a replacement part corresponding to the current to-be-played part, where a data amount of the to-be-adjusted data is the data adjustment amount; and playing the replacement part. With the technical solutions of the present invention, playback duration of a current to-be-played part can be adjusted in a manner of adding or deleting data, so that a controlling device and a controlled device generate sounds simultaneously when playing a to-be-played clip in a very simple implementation process.

Abstract: A method performed by radio network node for enabling channel handling of a channel between a wireless device and the radio network node in a wireless communication network. The channel is defined in continuous time and a sampling rate of the channel is non-uniform. The radio network node predicts a channel gain using a first sampling descriptor indicating a first momentary sampling frequency and a second sampling descriptor indicating a second momentary sampling frequency, wherein the first sampling descriptor operates on a different segment of continuous time than the second sampling descriptor. The predicted channel gain enables channel handling such as channel estimation and link adaptation.

Abstract: A method includes, but is not limited to, arranging several linear motors based on a 3D spherical model to form a linear motor matrix, and determining an azimuth of each linear motor on the 3D spherical model, performing information extraction for received 3D audio signal, and obtaining azimuth, frequency and amplitude information of the 3D audio signal reaching a central position of the 3D spherical model, generating a motor drive signal according to the azimuth, frequency and amplitude information of the 3D audio signal reaching a central position of the 3D spherical model, and driving a linear motor in the linear motor matrix within a certain scope away from the azimuth of the 3D audio signal to vibrate.

Abstract: Provided is a method of processing an audio signal, the method performed by a device and including dividing the audio signal into a main signal and a background signal; obtaining position information about a main speaker and an auxiliary speaker; mixing the main signal and the background signal, based on the position information; and outputting the mixed main signal and the mixed background signal to the main speaker and the auxiliary speaker, respectively.

Abstract: Embodiments relate to obtaining filter coefficients for a binaural synthesis filter; and applying a compensation filter to reduce artifacts resulting from the binaural synthesis filter; wherein the filter coefficients and compensation filter are configured to be used to obtain binaural audio output from a monaural audio input. The filter coefficients and compensation filter may be applied to the monaural audio input to obtain the binaural audio output. The compensation filter may comprise a timbre compensation filter.

Abstract: A method including, identifying at least one object of interest (OOI), determining a plurality of microphones capturing sound from the at least one OOI, determining, for each of the plurality of microphones, a volume around the at least one OOI, determining a spatial audio volume based on associating each of the plurality of microphones to the volume around the at least one OOI, and generating a spatial audio scene based on the spatial audio volume for free-listening-point audio around the at least one OOI.

Abstract: The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor.

Abstract: A automatic variable frequency electrolarynx with a microcontroller that generates a random but controlled frequency that creates a random tone. The controlled random tone allows the user of the electrolarynx to use the device without the monotone of standard electrolarynx. The electrolarynx also includes an encoder that allows the user to easily adjust the base tone of the device. The electrolarynx also includes an integrated LED light in the on/off switch to indicate whether the encoder is in volume control or tone control mode. The microcontroller also conducts an automatic power source check, and produces a musical tone when the battery is low.

Abstract: The present disclosure relates to reverberation generation for headphone virtualization. A method of generating one or more components of a binaural room impulse response (BRIR) for headphone virtualization is described. In the method, directionally-controlled reflections are generated, wherein directionally-controlled reflections impart a desired perceptual cue to an audio input signal corresponding to a sound source location. Then at least the generated reflections are combined to obtain the one or more components of the BRIR. Corresponding system and computer program products are described as well.

Abstract: In an audio processing system a center point stereo sound processor in combination with a digital transducer alignment sound processor converts regular left and right stereo audio source signals to sum and difference signals, enabling a center stage spatialized stereo acoustic image to be created directly from a forward-directed loudspeaker unit and a sideways-directed dipole loud speaker unit. The forward-directed loudspeaker unit includes one vertically-mounted loudspeaker. The sideways-directed dipole loudspeaker unit is located directly below and to the sides of the forward-directed loudspeaker unit and includes two speaker cones connected in parallel and wired out of phase with each other. The center point stereo sound processor receives a left and right, stereo, audio signal source and converts the left and right signals into sum and difference signals. The sum, left plus right signal drives the forward-directed loudspeaker unit.

Abstract: Left and right ear channels are determined for an end user. Each channel includes a head related transfer function (HRTF) filter to localize sound tailored to the geometry of the user's head and a frequency response filter tailored to the user's measured frequency response (hearing capability).