Abstract: An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.

Abstract: In general, techniques are described for specifying spherical harmonic coefficients in a bitstream. A device comprising one or more processors may perform the techniques. The processors may be configured to identify, from the bitstream, a plurality of hierarchical elements describing a sound field that are included in the bitstream. The processors may further be configured to parse the bitstream to determine the identified plurality of hierarchical elements.

Abstract: Methods and systems for fitting a bone conduction device are provided herein. These methods and systems comprise obtaining dynamic range parameters for a bone conduction device. These dynamic range parameters may include threshold and a maximum comfort levels for the bone conduction device. Once determined, the bone conduction device may use the dynamic range parameters in applying stimulation to a recipient.

Abstract: A processor-implemented method for spatial sound characterization is described. In one implementation, each of a plurality of source signals detected by a plurality of sensing devices, is segmented into a plurality of time frames. For each time frame, time-frequency transform of the source signals is derived, an estimated number of sources and at least one estimated direction of arrival corresponding to each of the source signals is obtained. Further, source signals are extracted by spatial separation based at least on the estimated directions of arrival and the estimated number of sources, and separated source signals are processed to yield a reference signal and side information.

Abstract: A method of processing a sound signal includes obtaining a phase difference or a time difference between sound signals received at both ears; determining a level difference between the sound signals based on the phase difference or time difference; determining gains of the sound signals to be output to both ears based on the level difference; and outputting the sound signals based on the determined gains.

Abstract: Systems and methods to calibrate listening devices are disclosed herein. In some embodiments, a method to calibrate earphones includes determining a Head Related Transfer Functions (HRTF) corresponding to different parts of a user's anatomy (e.g., one or both of a listener's pinnae). The resulting HRTFs are combined to form a composite HRTF. In some embodiments, a first and a second HRTF are respectively determined for a first and second part of the user's anatomy. A composite HRTF of the user is generated by combining portions of the first and second HRTFs.

Abstract: The invention relates to a loudspeaker system as well as a loudspeaker with controllable directivity as well as a method for controlling the individual sound transducer units in order to obtain a given target directivity. The loudspeaker comprises a plurality of sound transducer units distributed over two or more surfaces of a body. Each of the sound transducer units are utilized for frequency dependent a) level/gain adjustment and b) delay/phase adjustment of each individual sound transducer unit. The sound emitting transducer units are organized in two or more transducer arrays where the physical placement of transducer units included in the at least two or more transducer unit arrays are symmetrical organized and mirrored around a common axis.

Abstract: This disclosure describes techniques for coding of higher-order ambisonics audio data comprising at least one higher-order ambisonic (HOA) coefficient corresponding to a spherical harmonic basis function having an order greater than one. This disclosure describes techniques for adjusting HOA soundfields to potentially improve spatial alignment of the acoustic elements to the visual component in a mixed audio/video reproduction scenario. In one example, a device for rendering an HOA audio signal includes one or more processors configured to render the HOA audio signal over one or more speakers based on one or more field of view (FOV) parameters of a reference screen and one or more FOV parameters of a viewing window.

Abstract: A sound output device according to the present invention comprises a plurality of sub-sound outputting devices which are combined with or separated from each other and are connected to each other electrically or through a short range communication network; and a controller being mounted in one of the plurality of sub-sound outputting devices. The controller can determine the type of a sound channel to realize based on how the plurality of sub-sound outputting devices are combined and disposed and determine the roles of the plurality of sub-sound outputting devices in the determined sound channel.

Abstract: A system for generating and outputting one or more audio output signals has a decomposition module, a signal processor, and an output interface. The decomposition module can receive two or more audio input signals, to generate a direct component signal, having direct signal components of the audio input signals, and to generate a diffuse component signal, having diffuse signal components of the audio input signals. The signal processor can receive the direct component signal, the diffuse component signal and direction information, to generate one or more processed diffuse signals depending on the diffuse component signal, for each of the one or more audio output signals, to determine, depending on the direction of arrival, a direct gain, to apply the direct gain on the direct component signal to obtain a processed direct signal, and to combine the processed direct signal and a processed diffuse signal to generate the audio output signal.

Abstract: An example implementation involves a first device receiving, from a sensor, first location data that is associated with the first device, and, from a second device, second location data that is associated with the second device. The first device determines a first location of a listener relative to the first device based on the first and second location data. The first device generates a first sound field during media content playback based on the determined first location. The first device receives, from the sensor, third location data that is associated with the first device, and, from the second device, fourth location data associated with the second device. Based on the third and fourth location data, the first device determines a second location of the listener relative to the first device. Based on the second determined location, the first device generates a second sound field during media playback.

Abstract: Method and devices for providing surround audio signals are provided. Surround audio signals are received and are binaurally filtered by at least one filter unit. In some embodiments, the input surround audio signals are also processed by at least one equalizing unit. In those embodiments, the binaurally filtered signals and the equalized signals are combined to form output signals.

Abstract: A sound enhancement system (SES) that can enhance reproduction of sound emitted by headphones and other sound systems is disclosed. The SES improves sound reproduction by simulating a desired sound system without including unwanted artifacts typically associated with simulations of sound systems. The SES facilitates such improvements by transforming sound system outputs through a set of one or more binaural rendering filters derived from direct and indirect head-related transfer functions (HRTFs). Parameters of the binaural rendering filters are updated based on the head tracking angle of user wearing the headphones to render a stable stereo sound image. The head tracking angle may be determined from sensor data obtained from a digital gyroscope mounted in a headphone assembly.

Abstract: Decoding of Ambisonics representations for a stereo loudspeaker setup is known for first-order Ambisonics audio signals. But such first-order Ambisonics approaches have either high negative side lobes or poor localization in the frontal region. The invention deals with the processing for stereo decoders for higher-order Ambisonics HOA. The desired panning functions can be derived from a panning law for placement of virtual sources between the loudspeakers. For each loudspeaker a desired panning function for all possible input directions at sampling points is defined. The panning functions are approximated by circular harmonic functions, and with increasing Ambisonics order the desired panning functions are matched with decreasing error. For the frontal region between the loudspeakers, a panning law like the tangent law or vector base amplitude panning (VBAP) are used. For the rear directions panning functions with a slight attenuation of sounds from these directions are defined.

Abstract: An apparatus for generating a stereo output signal includes a manipulation information generator being adapted to generate manipulation information depending on a first signal indication value of a first input channel and on a second signal indication value of a second input channel, and a manipulator for manipulating a combination signal based on the manipulation information to obtain a first manipulated signal as a first output channel and a second manipulated signal as a second output channel. The combination signal is a signal derived by combining the first input channel and the second input channel. Furthermore, the manipulator is configured for manipulating the combination signal in a first manner, when the first signal indication value is in a first relation to the second signal indication value, or in a different second manner, when the first signal indication value is in a different second relation to the second signal indication value.

Abstract: It is disclosed inter alia a method comprising: estimating a value of entropy for a multi-channel audio signal; determining a channel configuration of the multi-channel audio signal from the value of entropy; and encoding the multi-channel audio signal, wherein the mode of encoding is dependent on the channel configuration.

Abstract: The present invention relates to an audio signal processing method, comprising the steps of: receiving a bit-stream containing a normal channel signal and an exceptional channel signal; decoding the normal channel signal and the exceptional channel signal from the received bit-stream; generating correlation information using the decoded normal channel signal and the decoded exceptional channel signal; generating a gain value by at least one of a first downmix method applying the same downmix gain value using the correlation information and a second downmix method applying variable gain values over time; and outputting the exceptional channel signal as a plurality of channel signals using the gain value.

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: An audio system that maintains an identical or similar direct-to-reverberant ratio for sound produced from a first speaker array and sound produced by a second speaker array at the location of a listener is described. The audio system may determine characteristics of the first and second speaker arrays, including the distance between the first speaker array and the listener and the second speaker array and the listener. Based on these characteristics, beam patterns are selected for one or more of the speaker arrays such that sound produced by each of the speaker arrays maintains a preferred direct-to-reverberant ratio at the location of the listener.

Abstract: A system and method is provided for selecting an audio tuning profile to apply to audio signals to generate a sound field acoustically optimized at least at one listening position in a listening environment, such as a vehicle passenger compartment. Each audio tuning profile may include a number of audio settings to be applied to an audio signal at one or more audio loudspeaker channels. The audio tuning profile may be selected based on the content or the source of the audio data signals. Thus, audio tuning may be based on the context of the audio.

Abstract: [Object] To provide a feeling of more openness to the user. [Solution] Provided is an information processing device, including: a listening environment characteristic information acquiring unit configured to acquire listening environment characteristic information indicating a characteristic of a listening environment based on an external sound collected through at least one microphone; and a musical signal processing unit configured to perform filtering of a musical signal according to a filter characteristic based on the acquired listening environment characteristic information.

Abstract: In one example, a method includes obtaining an audio signal comprising a plurality of elements; generating a first Higher-Order Ambisonics (HOA) soundfield that represents the audio signal; selecting a set of elements of the audio signal for encoding in a non-Higher-Order Ambisonics (HOA) domain; generating, based on the selected set of elements and a set of spatial positioning vectors, a second HOA soundfield that represents the selected set of elements; generating a third HOA soundfield that represents a difference between the first HOA soundfield and the second HOA soundfield; and generate a coded audio bitstream that includes a representation of the selected set of elements in the non-HOA domain, an indication of the set of spatial positioning vectors, and a representation of the third HOA soundfield.

Abstract: In general, techniques are described for obtaining one or more first vectors describing distinct components of a soundfield and one or more second vectors describing background components of the soundfield, both the one or more first vectors and the one or more second vectors generated at least by performing a transformation with respect to a plurality of spherical harmonic coefficients.

Abstract: A system comprising a video source, one or more audio sources and a computing device. The video source may be configured to generate a plurality of video streams that capture a view of an environment. The one or more audio sources may be configured to capture audio data of the environment. The computing device may comprise one or more processors configured to (i) perform a stitching operation on the plurality of video streams to generate a video signal representative of an immersive field of view of the environment, (ii) generate a sound field based on the audio data, (iii) identify an orientation for the sound field with respect to the video signal, and (iv) determine a rotation of the sound field based on the orientation. The rotation of the sound field aligns the sound field to the video signal.

Abstract: A method of generating a multi-channel audio signal includes: representing locations of a plurality of speakers as a plurality of polygons whose vertices are located at locations of corresponding speakers; acquiring a location of an object sound; calculating distances between the plurality of polygons and the location of the object sound; selecting one of the plurality of polygons on the basis of the calculated distances; and generating a multi-channel audio signal that corresponds to speakers corresponding to the selected polygon by mapping the object sound to the speakers corresponding to the selected polygon.

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: An apparatus and a method for enhancing a spatial perception of an audio signal are provided creating increased interaural-level differences. To obtain this effect, two dipoles are used: one for producing a left audio signal and one for producing a right audio signal.

Abstract: According to a sound image direction sense processing method and apparatus, a left-ear channel signal, a right-ear channel signal, and a centered channel signal that are of a sound source are obtained; whether a direction of the sound source is a front direction is determined according to the left-ear channel signal, the right-ear channel signal, and the centered channel signal; and when the direction of the sound source is the front direction, at least one type of the following processing: front direction enhancing processing or rear direction weakening processing is performed separately on the left-ear channel signal and the right-ear channel signal. Therefore, a difference between front direction sense and rear direction sense of a sound image may be enlarged, so that accuracy of determining a direction of a sound source may be improved.

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: One embodiment of the present invention provides technology which enables a user to process non-compressed input content or compressed input content according to settings of the user, and technology capable of selectively supporting adding, editing, and eliminating an object from the compressed input content on the basis of various coding methods.

Abstract: An augmented reality display system comprises a passable world model data comprises a set of map points corresponding to one or more objects of the real world. The augmented reality system also comprises a processor to communicate with one or more individual augmented reality display systems to pass a portion of the passable world model data to the one or more individual augmented reality display systems, wherein the piece of the passable world model data is passed based at least in part on respective locations corresponding to the one or more individual augmented reality display systems.

Abstract: A hearing system includes one or more hearing devices configured to be worn by a user. Each hearing device includes a signal source that provides an input electrical signal representing a sound of a virtual source. A filter implements a head related transfer function (HRTF) to add spatialization cues associated with a virtual location of the virtual source to the electrical signal and outputs a filtered electrical signal that includes the spatialization cues. A speaker of the hearing device converts the filtered electrical signal into an acoustic signal and plays the acoustic signal to the user. The system includes motion tracking circuitry that tracks motion of the user as the user moves in a direction of a perceived location that the user perceives to be the virtual location of the virtual source. Head related transfer function (HRTF) individualization circuitry determines a difference between the virtual location and the perceived location in response to the motion of the user.

Abstract: Implementations disclose a mesh network of nearby mobile devices as a combined speaker system for audio. A method includes detecting one or more other computing devices within a geographic vicinity of a computing device, transmitting a confirmation to join a speaker mesh network with the detected one or more other computing devices, receiving a playback synchronization instruction for a media item, adjusting playback of the media item according to the playback synchronization instruction, wherein the one or more other computing devices in the speaker mesh network provide synchronized playback of the media item according to the playback synchronization instruction, receiving a stereo sound and audio equalization instruction, and applying, according to the stereo sound and audio equalization instruction, stereo sound and audio equalization settings to the playback of the media item at the computing device.

Abstract: Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal.

Abstract: Methods and systems for audio source separation in real-time are described. In an embodiment, the present disclosure describes reading and decoding an audio source into PCM samples, fragmenting Pulse Code Modulation (PCM) samples into fragments, transforming fragments into spectrograms, performing audio source separation using a training database that includes a training dictionary and non-negative matrix factorization (NMF) to generate a set of component signals, and streaming the component signals to a playback engine. In an embodiment, a semantic equalizer graphical user allows for fading of individual component signals.

Abstract: A method and a device for operating an arrangement of sound transducers according to the wave-field synthesis principle. In order to supply an extended audience region with the same signal, the same signal content is generated by at least two virtual sound sources, which are arranged such that the wavefronts thereof are directed only onto a part audience area, rather than generating only a single beam extending over the entire audience area. The wavefronts of the distributed virtual sound sources add up vectorially in the plane of the arrangement of sound transducers, whereby the effectiveness of the sound generation is increased.

Abstract: Systems, methods, and computer-readable storage media for generating an immersive three-dimensional sound space for searching audio. The system generates a three-dimensional sound space having a plurality of sound sources playing at a same time, wherein each of the plurality of sound sources is assigned a respective location in the three-dimensional sound space relative to one another, and wherein a user is assigned a current location in the three-dimensional sound space relative to each respective location. Next, the system receives input from the user to navigate to a new location in the three-dimensional sound space. Based on the input, the system then changes each respective location of the plurality of sound sources relative to the new location in the three-dimensional sound space.

Abstract: An audio signal processing device for generating a plurality of output signals for a plurality of loudspeakers from an input audio signal comprises a driving function determining unit adapted to determined driving functions of a plurality of loudspeakers for generating a virtual left binaural signal source and a virtual right binaural signal source based upon a position and a directivity of the virtual left binaural signal source, a position and a directivity of the virtual right binaural signal source and positions of the plurality of loudspeakers. Moreover, it comprises a filtering unit adapted to filter a left binaural signal and a right binaural signal using the driving functions of the plurality of loudspeakers resulting in the plurality of output signals. The left binaural signal and the right binaural signal constitute the input audio signal or are derived there from.

Abstract: Example methods and systems for displaying one or more indications that indicate (i) the direction of a source of sound and (ii) the intensity level of the sound are disclosed. A method may involve receiving audio data corresponding to sound detected by a wearable computing system. Further, the method may involve analyzing the audio data to determine both (i) a direction from the wearable computing system of a source of the sound and (ii) an intensity level of the sound. Still further, the method may involve causing the wearable computing system to display one or more indications that indicate (i) the direction of the source of the sound and (ii) the intensity level of the sound.

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: 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: Systems and methods for stereo separation and directional suppression are provided. An example method includes receiving a first audio signal, representing sound captured by a first microphone associated with a first location, and a second audio signal, representing sound captured by a second microphone associated with a second location. The microphones comprise omni-directional microphones. The distance between the first and second microphones is limited by the size of a mobile device. A first channel signal of a stereo signal is generated by forming, based on the first and second audio signals, a first beam at the first location. A second channel signal of the stereo signal is generated by forming, based on the first and second audio signals, a second beam at the second location. First and second directions, associated respectively with the first and second beams, are fixed relative to a line between the first and second locations.

Abstract: An audio reproduction system for reproducing audio data of at least one audio object and/or at least one sound source of an acoustic scene in a given environment comprising: at least two audio systems acting distantly apart from each other, wherein one of the audio systems is adapted to reproduce the audio object and/or the sound source in a first distance range to a listener and another of the audio systems is adapted to reproduce the audio object and/or the sound source in a second distance range to the listener, wherein the first and second distance ranges are different and possibly spaced apart from each other or placed adjacent to each other; and a panning information provider adapted to process at least one input to generate at least one panning information for each audio system to drive the at least two audio systems.

Abstract: A signal processor includes an input unit that receives a first audio signal and a second audio signal including mutually correlated components, a delay unit that delays the first audio signal received at the input unit by a prescribed delay time, a synthesis unit that synthesizes the first audio signal having been delayed by the delay unit with the second audio signal received at the input unit, and outputs a third audio signal resulting from synthesis, and a frequency band restriction unit that restricts a level of the first audio signal before the synthesis in a prescribed frequency band including a frequency of a dip occurring at a lowest frequency among a plurality of dips occurring in a frequency characteristic of the third audio signal as a result of the synthesis performed by the synthesis unit.

Abstract: A method selects a location where to localize binaural sound to a listener. Sounds are assigned to different zones or different sound localization points (SLPs). The sounds are convolved with sound localization information so the sounds externally localize as binaural sound away from the listener and into the assigned zone or to the assigned SLP.

Abstract: An audio processing system, such as an upmixer, may be capable of separating diffuse and non-diffuse portions of N input audio signals. The upmixer may be capable of detecting instances of transient audio signal conditions. During instances of transient audio signal conditions, the up-mixer may be capable of adding a signal-adaptive control to a diffuse signal expansion process in which M audio signals are output. The upmixer may vary the diffuse signal expansion process over time such that during instances of transient audio signal conditions the diffuse portions of audio signals may be distributed substantially only to output channels spatially close to the input channels. During instances of non-transient audio signal conditions, the diffuse portions of audio signals may be distributed in a substantially uniform manner.

Abstract: Encoding of Higher Order Ambisonics (HOA) signals commonly results in high data rates. A method for low bit-rate encoding frames of an input HOA signal having coefficient sequences comprises computing (s110) a truncated HOA representation (CT(k)), determining (s111) active coefficient sequences (IC,ACTT(k)), estimating (s16) candidate directions (MDIR(k)), dividing (s15) the input HOA signal into a plurality of frequency subbands (f1, . . . , fF), estimating (s161) for each of the frequency subbands a subset of candidate directions (MDIR(k)) as active directions (MDIR(k,f1), . . . , MDIR(k,fF)) and for each active direction a trajectory, computing (s17) for each frequency subband directional subband signals from the coefficient sequences of the frequency subband according to the active directions, calculating (s18) for each frequency subband a prediction matrix (A(k,f1), . . .

Abstract: A method, system, and apparatus including data comprising an identification of content played by playback systems and at least one metric associated with the playback systems. Determining a characteristic common to the playback systems and generating data comprising a representation of the at least one characteristic.

Abstract: Systems, methods, and apparatus for backward-compatible coding of a set of basis function coefficients that describe a sound field are presented.

Abstract: Techniques are disclosed for synchronizing gain adjustments across a cascaded network of audio gain stages having variant operating delays. In particular, a delay-synchronized volume adjustment system configured in accordance with an embodiment of the present disclosure includes a controller operatively coupled to the cascaded network of audio and configured to apply gain adjustments in a synchronized manner that accounts for operating delays that are inherent to each gain stage. In an embodiment, the controller synchronously adjusts each gain stage relative to a corresponding operating delay such that gain adjustments fully propagate at substantially a same point in time within a given acceptable tolerance, and thus, eliminates or otherwise mitigates perceivable volume shifts when mixing audio from two or more audio sources.