Abstract: When compressing an HOA data frame representation, a gain control (15, 151) is applied for each channel signal before it is perceptually encoded (16). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (?e) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalisation of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to ?e=?log2(?log2(?{square root over (KMAX)}·O)?+1)?.

Abstract: Provided is an information processing apparatus that is used by being adhered to the body of a user. The information processing apparatus includes a sound pickup sensor that has a sound pickup function, a communicating part that wirelessly transmits audio data picked up by the sound pickup sensor to the outside, a control part that controls the sound pickup sensor and the communicating part, a power source part that supplies a power source to at least one of the sound pickup sensor, the communicating part, or the control part, a housing part that accommodates therein at least one of the sound pickup sensor, the communicating part, the control part, or the power source part, and an adhering part that fixes the housing part to the user. The sound pickup part picks up a sound of the user using, for example, flesh conduction or bone conduction.

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 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: Disclosed is an audio signal processing device for processing an audio signal. The audio signal processing device includes a processor. The processor obtains an input audio signal including an object audio signal, selects at least one of a plurality of rendering methods based on an azimuth of a sound object with respect to a listener, corresponding to the object audio signal in a virtual space simulated by an output audio signal, renders the object audio signal using a selected rendering method, and outputs the output audio signal including the rendered object audio signal.

Abstract: A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is processed with an early part of the room impulse response separate from a late reverberation of the room impulse response, wherein the processing of the late reverberation has generating a scaled reverberated signal, the scaling being dependent on the audio signal. The processed early part of the audio signal and the scaled reverberated signal are combined.

Abstract: An information processing method includes obtaining an audio signal that represents sound collected by a sound collecting device; calculating, using the audio signal obtained, a volume of the sound collected by the sound collecting device; identifying, using the audio signal obtained, a type of a sound source of the sound collected by the sound collecting device; estimating a distance between the sound collecting device and the sound source of the sound collected by the sound collecting device based on a standard volume and the volume calculated, the standard volume being (i) stored in a database in which the type of the sound source and the standard volume are preassociated with each other, (ii) the volume of the sound from the sound source at a predetermined distance, and (iii) associated with the type of the sound source identified; and outputting an estimation result.

Abstract: A sound reproducing method used in sound reproducing apparatus that includes the steps outlined below is provided. A sound signal is generated with a 3D sound generating process according to listener data and sound data. Pre-recorded sound data is retrieved to further generate a target distance function corresponding to the sound distance. A fixed head-related transfer function corresponding to a fixed distance is retrieved. A target head-related transfer function corresponding to the sound distance is generated by adapting the target distance function to the fixed head-related transfer function. The sound signal is reproduced by multiplying the sound signal by the target head-related transfer function.

Abstract: An apparatus including: an input configured to receive from at least two microphones at least two audio signals; at least two processor instances configured to generate separate output audio signal tracks from the at least two audio signals from the at least two microphones; a file processor configured to link the at least two output audio signal tracks within a file structure.

Abstract: A voice providing method includes obtaining position information from a voice reproducing device, and performing processing, in a case in which a first position of obtained position information and a second position of a virtual talker are in a predetermined relationship, to cause the voice reproducing device to reproduce voice for the virtual talker to have a dialogue with a user with content according to the user.

Abstract: An expandable wireless speaker system and method is disclosed. The speaker system comprises a center module and at least two or more speakers. The center module has a unique identification number and has mechanical connections on at least one side. The center module is configured to receive an external device, wirelessly or directly connected and has a transceiver configured to scan for an unused channel and transmit audio. The speakers each have a mechanical connection on either side to connect to the center module or the other speakers. The two or more speakers are configured to scan the communication spectrum for the center module's unique identification and when the two or more speakers locate the center module's unique identification, the two or more speakers pair with the center module to receive audio. The communication spectrum can be a wireless channel such as 5.8 GHz or the FM radio spectrum.

Abstract: A method and apparatus to determine a direction of arrival (DOA) of a talker in the presence of a source of spatially-coherent noise. A time sequence of audio samples that include the spatially-coherent noise is received and buffered. Aided by previously known data, a trigger point is detected in the time sequence of audio samples when the talker begins to talk. The buffered time sequence of audio samples is separated into a noise segment and a signal-plus-noise segment based on the trigger point. For each direction of a plurality of distinct directions: an energy difference is computed for the direction between the noise segment and the signal-plus-noise segment, and the DOA of the talker is selected as the direction of the plurality of distinct directions having a largest of the computed energy differences.

Abstract: An input unit receives input of an assumed listening position of sound of an object, which is a sound source, and outputs assumed listening position information indicating the assumed listening position. A position information correction unit corrects position information of each object on the basis of the assumed listening position information to obtain corrected position information. A gain/frequency characteristic correction unit performs gain correction and frequency characteristic correction on a waveform signal of an object on the basis of the position information and the corrected position information. A spatial acoustic characteristic addition unit further adds a spatial acoustic characteristic to the waveform signal resulting from the gain correction and the frequency characteristic correction on the basis of the position information of the object and the assumed listening position information. The present technology is applicable to an audio processing device.

Abstract: A sound input/output device for a vehicle includes: microphones that are provided within a vehicle cabin; a plurality of directional speakers that are provided within the vehicle cabin, and that output sound to respective seats; an output position specifying section that specifies a seat that is an output destination from a voice collected by one or more of the microphones or from input from an utterer of a voice; and an output control section that outputs sound from one or more of the speakers to a seat specified by the output position specifying section.

Abstract: A system of crosstalk cancelled zone creation in audio playback comprising: main transducers emitting stereo soundwaves of an audio playback; a local system comprising at least two or more close-proximity-transducers (CPTs), each is arranged proximal to one of left and right-side ear canals of a listener. Each of the CPTs comprises: a position tracking device for tracking the relative positions of the main transducers to the CPT and the other CPTs; a control unit for receiving the relative position data from the position tracking device and generating control signal according to the relative position data for the generation of crosstalk cancellation (XTC) soundwaves. Each of the CPTs is configured to generate XTC soundwaves corresponding to the stereo soundwaves arriving at the corresponding ear of the listener. The generated XTC soundwaves are synchronized with the audio playback and with respect to the relative positions.

Abstract: Processing input audio channels for generating spatial audio can include receiving a plurality of microphone signals that capture a sound field. Each microphone signal can be transformed into a frequency domain signal. From each frequency domain signal, a direct component and a diffuse component can be extracted. The direct component can be processed with a parametric renderer. The diffuse component can be processed with a linear renderer. The components can be combined, resulting in a spatial audio output. The levels of the components can be adjusted to match a direct to diffuse ratio (DDR) of the output with the DDR of the captured sound field. Other aspects are also described and claimed.

Abstract: Methods, systems, and devices for head-related transfer function generation are described. A device may receive a digital representation of a first audio signal associated with a location relative to a subject, and select from a database a first reference head-related transfer function measurement pair corresponding to the location of the first audio signal. The device may then obtain a second head-related transfer function measurement pair by performing a style transfer operation on the selected reference head-related transfer function measurement pair based on a set of head-related transfer function measurement pairs specific to the subject. As a result, the device may output a second audio signal based on the digital representation of the first audio signal and the second head-related transfer function measurement pair.

Abstract: A method expedites playing sound of a talking emoji from a first person with a first portable electronic device (PED) to a second person with a second PED. The second PED receives the talking emoji in mono sound and convolves the mono sound into binaural sound before receiving a request to play the sound to the second user. The second PED then plays the sound of the talking emoji in binaural sound after receiving the request from the second user.

Abstract: A portable electronic device (PED) divides an area around a user into a three-dimensional (3D) zone. A wearable electronic device worn on a head of the user displays the 3D zone in response to the wearable electronic device detecting that the user is leaving the zone. The wearable electronic device plays binaural sound that emanates to the user from sound localization points (SLPs) inside the zone.

Abstract: Methods, systems, computer-readable media, and apparatuses for HRTF profile selection are presented. In one example, a device prompts a user to follow a simple procedure to obtain measurements that are matched to a suitable high-resolution HRTF profile.

Abstract: A computer-implemented method for calibrating audio of a virtual reality device, the method comprising: determining a difference between a perceived tone location and an actual audible tone location, in response to an emitting of the actual audible tone; wherein the tone comprises a spectral tone, and creating and calibrating a user ear model by: emitting the spectral tone at a random time during a simulation by the virtual reality device; determining the perceived tone location during the simulation; and computing an error adjustment for one or more geometric variables of a default ear model based on a result of the determining.

Abstract: Media content playback is based on an identified geographic location of a sound source. A geolocation of a target venue is determined. A first geolocation of a media playback device is determined. An indication that the media playback device is within a threshold distance of the target venue is received. Media content associated with the target venue is transferred to the media playback device, wherein one or more characteristics of the media content are dynamically modified to simulate audio originating from the target device. The one or more characteristics of the media content can include audio content characteristics associated with a timing filter, a loudness filter, and a timbre filter.

Abstract: The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.

Abstract: An audio system processes a multi-channel surround sound input audio signal into a stereo signal for left and right speakers, while preserving the spatial sense of the sound field of the input audio signal. A subband spatial processing is performed on a left input channel, a right input channel, a left peripheral input channel, and a right peripheral input channel of the input signal to create spatially enhanced channels. Binaural filters may be applied to the peripheral input channels or the spatially enhanced channels. Crosstalk cancellation is performed on the spatially enhanced channels to create a left crosstalk cancelled channel and a right crosstalk cancelled channel.

Abstract: Systems, methods, and computer readable media to improve the operation of an electronic device having multiple microphones organized in an arbitrary, but known, arrangement in the device are described (i.e., having a specific form-factor). In general, techniques are disclosed for using a priori knowledge of an electronic device's spatial acoustic transfer functions to recreate or reconstitute a prior recorded three-dimensional (3D) audio field or environment. More particularly, techniques disclosed herein enable the efficient recording of a 3D audio field. That audio field may later be reconstituted using an acoustic characterization based on the device's form-factor. In addition, sensor data may be used to rotate the audio field so as to enable generating an output audio field that takes into account the listener's head position.

Abstract: Disclosed is an apparatus for setting a target position using a GUI screen. The apparatus includes a display that displays a representation of a listener and a representation of a hemispherical dome with the listener at the center as projected on a horizontal plane. A user sets a radius of the hemispherical dome through a click operation of a mouse and sets a target position of a sound image with respect to the listener through a drag operation toward the center. The sound image position is represented by a dot O. The dot O represents a position of the sound image with respect to the listener in the left-to-right direction, in the front-to-rear direction, and in the height direction.

Abstract: A method including providing an input audio signal in a first path and applying an interpolated head-related transfer function (HRTF) pair based upon a direction to generate direction dependent first left and right signals in the first path; providing the input audio signal in a second path, where the second path includes a plurality of filters and a respective amplifier for each filter, where the amplifiers are configured to be adjusted based upon the direction, and applying to an output from each of the filters a respective head-related transfer function (HRTF) pair to generate direction dependent second left and right signals for each filter in the second path; and combining the generated left signals to form a left output signal for a sound reproduction, and combining the generated right signals to form a right output signal for the sound reproduction.

Abstract: A handheld portable electronic device (HPED) designates a sound localization point (SLP) for binaural sound. A digital signal processor (DSP) processes the sound with head-related transfer functions (HRTFs) to generate the binaural sound. A wearable electronic device (WED) displays a sphere that includes the SLP.

Abstract: Example techniques may involve performing aspects of a spatial calibration. An example implementation may include detecting a trigger condition that initiates calibration of a media playback system including multiple audio drivers that form multiple sound axes, each sound axis corresponding to a respective channel of multi-channel audio content The implementation may also include causing the multiple audio drivers to emit calibration audio that is divided into constituent frames, the multiple sound axes emitting calibration audio during respective slots of each constituent frame. The implementation may further include recording the emitted calibration audio. The implementation may include causing delays for each sound axis of the multiple sound axes to be determined, the determined delay for each sound axis based on the slots of recorded calibration audio corresponding to the sound axes and causing the multiple sound axes to be calibrated.

Abstract: An electronic device is provided.

Abstract: Providing a virtual acoustic environment comprises determining updates to audio signals based at least in part on information in sensor output, including, for each of multiple time intervals: determining an updated position of a wearable audio device, based at least in part on position information in the sensor output; determining layouts of at least four virtual walls, where the layouts are determined such that the updated position is within a space defined by the virtual walls; determining positions of at least four image audio sources associated with a virtual audio source, where a position of each image audio source is dependent on a layout of a corresponding one of the virtual walls and a position of the virtual audio source; and processing the audio signals using an update determined based at least in part on the respective positions of the virtual audio source and the image audio sources.

Abstract: The present application provides method and apparatus for a binaural hearing assistance system using a monaural audio signal input. The system, in various examples, provides adjustable delay/phase adjustment and sound level adjustment. Different embodiments are provided for receiving the monaural signal and distributing it to a plurality of hearing assistance devices. Different relaying modes are provided. Special functions are supported, such as telecoil functions. The system also has examples that account for a head-related transfer function in providing advanced sound processing for the wearer. Other examples are provided that are described in the detailed description.

Abstract: The invention relates to an audio system (2) for disposing in the external region of an ear, having a carrier element (4) for affixing to the head of the user and comprising at least one hollow space (26, 27, 28) having a primary opening (11, 12, 13) oriented in the direction of an ear opening (22) of the intended ear when the carrier element (4) is disposed as intended, and having at least one speaker (5, 6, 7) disposed in the hollow space (26, 27, 28), the cavity (14, 15, 16) thereof being at least partially formed by a first hollow space region (29) and by means of which first sound waves (35) can be emitted through the primary opening (11, 12, 13) in the direction of the ear spaced apart from said opening.

Abstract: Methods, systems, computer-readable media, and apparatuses for characterizing portions of a soundfield are presented. Some configurations include estimating a total energy of a soundfield associated with a scene space and, for each of at least some of a plurality of regions of the scene space, estimating an energy of a portion of the soundfield that corresponds to the region and creating a corresponding metadata field that indicates a location of the region within the space and a relation between the estimated total energy and the estimated energy that corresponds to the region.

Abstract: Techniques for optimizing a player based on the addition of a second player are disclosed. In an embodiment, when a first player no longer needs to play certain audio frequencies due to the addition of a second player, the gain of the first player is automatically increased as part of the setup process. In another embodiment, when a first player needs to play certain audio frequencies, for example due to the removal of a second player, the gain of the first player is automatically decreased. Many other embodiments are disclosed.

Abstract: An audio system generates virtual acoustic environments with three-dimensional (3-D) sound from legacy video with two-dimensional (2-D) sound. The system relocates sound sources within the video from 2-D to into a 3-D geometry to create an immersive 3-D virtual scene of the video that can be viewed using a headset. Accordingly, an audio processing system obtains a video that includes flat mono or stereo audio being generated by one or more sources in the video. The system isolates the audio from each source by segmenting the individual audio sources. Reverberation is removed from the audio from each source to obtain each source's direct sound component. The direct sound component is then re-spatialized to the 3-D local area of the video to generate the 3-D audio based on acoustic characteristics obtained for the local area in the video.

Abstract: Disclosed is a method for determining an audio filter. The method includes obtaining head-tracking data representing a head pose, also referred to as a head position, relative to a torso pose of an intended subject, and selecting a representation of a Head Related Transfer Function, HRTF, from a HRTF database, based on the head-tracking data and information representative of the position and/or direction of a sound source. The method also includes determining the audio filter based on the selected HRTF representation.

Abstract: An apparatus for estimating an inter-channel time difference between a first channel signal and a second channel signal, includes: a calculator for calculating a cross-correlation spectrum for a time block from the first channel signal in the time block and the second channel signal in the time block; a spectral characteristic estimator for estimating a characteristic of a spectrum of the first channel signal or the second channel signal for the time block; a smoothing filter for smoothing the cross-correlation spectrum over time using the spectral characteristic to obtain a smoothed cross-correlation spectrum; and a processor for processing the smoothed cross-correlation spectrum to obtain the inter-channel time difference.

Abstract: The present technology relates to an audio processing apparatus and method and a program that are configured to adjust audio characteristics more easily. This audio processing apparatus has a display control unit configured, on the basis of object positional information of an audio object, to cause a display unit to display an audio object information image indicative of a position of the audio object and a selection unit configured to select the predetermined audio object from among one or a plurality of the audio objects. The present technology is applicable to video/audio processing apparatuses.

Abstract: An apparatus for mapping a first input channel and a second input channel of an input channel configuration to at least one output channel of an output channel configuration, wherein each input channel and each output channel has a direction in which an associated loudspeaker is located relative to a central listener position, configured to map the first input channel to a first output channel of the output channel configuration; and despite of the fact that an angle deviation between a direction of the second input channel and a direction of the first output channel is less than an angle deviation between a direction of the second input channel and the second output channel and/or is less than an angle deviation between the direction of the second input channel and the direction of the third output channel, map the second input channel to the second and third output channels by panning between the second and third output channels.

Abstract: Examples are disclosed relating to providing spatialized audio to multiple users. In one example, a computing device presents spatialized audio to multiple users within an environment via communicative connection to one or more wearable spatial audio output devices. For each communicatively connected wearable spatial audio output device, a user-specific subset of audio tracks is generated from a set of audio tracks for a dynamic audio object positioned within the environment based on one or more user-specific parameters. A location of the wearable spatial audio output device is determined relative to the dynamic audio object, and based upon this location, a device-specific spatialized audio mix is generated that includes the user-specific subset of audio tracks. The device-specific spatialized audio mixes are sent to the wearable spatial output devices, and playback of the device-specific spatialized audio mixes are synchronously initiated at each wearable spatial audio output device.

Abstract: The present invention relates to a method and an apparatus for processing a signal, which are used to effectively reproduce an audio signal, and more particularly, to a method for generating a filter for an audio signal, which are used for implementing a filtering for input audio signals with a low computational complexity and a parameterization apparatus therefor.

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: A method for controlling a scene sound effect and related products are disclosed. The method comprises: a monitoring service is enabled after electronic equipment is turned on; the electronic equipment monitors a newly created view in the electronic equipment by means of the monitoring service to determine whether there is the newly created view in the electronic equipment, a characteristic of the newly created view having a mapping relationship with an application in the electronic equipment; when the electronic equipment determines that there is the newly created view in the electronic equipment, the electronic equipment determines the application having the mapping relationship with the characteristic of the newly created view in the electronic equipment according to the mapping relationship; and the electronic equipment acquires a scene sound effect corresponding to the application and sets a sound effect of the electronic equipment as the scene sound effect.

Abstract: An earpiece includes an earpiece housing, wherein the earpiece housing is configured to substantially encompass an opening of an ear canal to physically block ambient sound from entering the ear canal. The earpiece further includes a processor disposed within the earpiece housing, at least one microphone operatively connected to the processor and configured to sense ambient sound, and a speaker operatively connected to the processor. The processor is configured to process the ambient sound from the microphone and add additional ambient sound to produce a modified ambient sound at the speaker.

Abstract: A signal processing apparatus and method are provided for separating a plurality of mixture signals from a MIMO system to iteratively obtain a plurality of output signals. The plurality of mixture signals are a response of the MIMO system to a plurality of source signals. The signal processing apparatus comprises a plurality of blind source separators including a first blind source separator based on a first blind source separation technique or algorithm and a second blind source separator based on a second blind source separation technique or algorithm, wherein the first blind source separator is configured to compute a first plurality of preliminary output signals on the basis of a first set of coefficients describing the MIMO system and wherein the second blind source separator is configured to compute a second plurality of preliminary output signals on the basis of a second set of coefficients describing the MIMO system.

Abstract: A signal processing device includes a prior distribution storage that stores a prior distribution group that is a set of prior distributions representing representative spectral shapes of spectral bases of an acoustic event specified as a detection target, and a basis generation unit that, using as input a spectrogram for basis generation, generates an acoustic event basis group that is a set of the spectral bases of the acoustic event specified as a detection target. The basis generation unit performs sound source separation on the spectrogram for basis generation and thereby generates an acoustic event basis group, using respective prior distributions included in a prior distribution group as a constraint for spectral bases of a corresponding acoustic event.

Abstract: Methods, systems, and computer readable media for simulating sound propagation are disclosed. According to one method, the method includes decomposing a virtual environment scene including at least one object into a plurality of surface regions, wherein each of the surface regions includes a plurality of surface patches. The method further includes organizing sound rays generated by a sound source in the virtual environment scene into a plurality of path tracing groups, wherein each of the path tracing groups comprises a group of the rays that traverses a sequence of surface patches. The method also includes determining, for each of the path tracing groups, a sound intensity by combining a sound intensity computed for a current time with one or more previously computed sound intensities respectively associated with previous times and generating a simulated output sound at a listener position using the determined sound intensities.

Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: Audio data processor, having: a receiver interface for receiving encoded audio data and metadata related to the encoded audio data; a metadata parser for parsing the metadata to determine an audio data manipulation possibility; an interaction interface for receiving an interaction input and for generating, from the interaction input, interaction control data related to the audio data manipulation possibility; and a data stream generator for obtaining the interaction control data and the encoded audio data and the metadata and for generating an output data stream, the output data stream having the encoded audio data, at least a portion of the metadata, and the interaction control data.