Abstract: Virtualizing speakers for a headphone set can include determining a location of a display. Locations of one or more virtual speakers can be assigned based on the location of the display. A first virtual speaker can be located at the display. A position of a head of a user can be tracked. Audio content can be spatialized with a spatial renderer to generate spatialized audio signals, based on the tracked position of the head and the locations of the virtual speakers. Other aspects are also described and claimed.

Abstract: A loudspeaker system (200) includes an overhead sound image generating (e.g., ATMOS™) elevation sound projecting loudspeaker transducer or array (210) for reproducing an elevation signal and a second cancellation loudspeaker or transducer array (250) for generating and projecting a direct signal cancellation. The cancellation speaker or array (250) is driven with a filtered, polarity reversed version of the elevation signal to cancel undesired direct sound (160) from elevation speaker (210) which would otherwise diminish the quality of elevation signal reproduction for a listener L.

Abstract: An intelligent hub for interfacing with other devices and performing smart audio or video source selection.

Abstract: A computer-implemented method for generating a device-specific room impulse response (DSRIR) describing an acoustic characteristic of a device and a room as received by the device includes: generating an at least first device-related transfer function (DRTF), wherein the at least first device related transfer function describes the acoustic characteristic of the device as received by the at least first microphone; generating a spatial room impulse response (SRIR), wherein the spatial room impulse response describes the acoustic characteristic of the room from at least one room sound source in the room and received at an at least one listening point in the room from at least one direction; and generating the device specific room impulse response (DSRIR) by combining the device related transfer function and the spatial room impulse response.

Abstract: An apparatus including circuitry configured to: obtain a binaural audio signal; obtain, based on the binaural audio signal, at least one direction parameter of at least one frequency band of the binaural audio signal; process the binaural audio signal to generate at least two audio signals for loudspeaker reproduction by modifying an inter-channel difference of the at least one frequency band of the binaural audio signal based on the at least one direction parameter for the at least one frequency band; and output the at least two audio signals for loudspeaker reproduction.

Abstract: An acoustic image source model for early reflections in a room is generated by iteratively mirroring (305) rooms around boundaries (e.g. walls) of rooms of the previous iteration. The boundaries around which to mirror in each iteration is determined (303) by a specific selection criterion including requiring that mirror directions cannot be reversed, cannot be in an excluded direction and cannot be repeated unless in a continuous series of mirrorings.

Abstract: Described herein is a method (30) of rendering an audio signal (17) for playback in an audio environment (27) defined by a target loudspeaker system (23), the audio signal (17) including audio data relating to an audio object and associated position data indicative of an object position. Method (30) includes the initial step (31) of receiving the audio signal (17). At step (32) loudspeaker layout data for the target loudspeaker system (23) is received. At step (33) control data is received that is indicative of a position modification to be applied to the audio object in the audio environment (27). At step (38) in response to the position data, loudspeaker layout data and control data, rendering modification data is generated. Finally, at step (39) the audio signal (17) is rendered with the rendering modification data to output the audio signal (17) with the audio object at a modified object position that is between loudspeakers within the audio environment (27).

Abstract: A method and apparatus for rendering a volume sound source are disclosed. The method of rendering a volume sound source may include identifying information about a listener and information about the volume sound source, determining a corresponding area in which a source element is disposed in the volume sound source in consideration of the information about the listener, determining an angle between the listener and the corresponding area based on the information about the listener and the information about the volume sound source, determining a number of source elements disposed in the corresponding area according to the angle, determining a position and a gain of the source element using i) the number of source elements and ii) a distance between the listener and the volume sound source, and rendering the volume sound source according to the position and the gain of the source element.

Abstract: An audio processing apparatus comprises a receiver (705) which receives audio data including audio components and render configuration data including audio transducer position data for a set of audio transducers (703). A renderer (707) generating audio transducer signals for the set of audio transducers from the audio data. The renderer (7010) is capable of rendering audio components in accordance with a plurality of rendering modes. A render controller (709) selects the rendering modes for the renderer (707) from the plurality of rendering modes based on the audio transducer position data. The renderer (707) can employ different rendering modes for different subsets of the set of audio transducers the render controller (709) can independently select rendering modes for each of the different subsets of the set of audio transducers (703).

Abstract: The present disclosure relates to an image display apparatus and a method thereof. The image display apparatus according to an embodiment of the present disclosure comprises: a display; an external device interface configured to perform short-range communication; and a controller configured to: when the number of external devices communicatively connected with the image display apparatus is one, or when a bandwidth usage for the short-range communication is less than a predetermined reference, transmit an entire signal to the one external device, when the number of the external devices is two or more, and the bandwidth usage is greater than or equal to the predetermined reference, acquire a plurality of channel signals from the entire signal; transmit a first channel signal to a first external device among the external devices; and transmit a second channel signal to a second external device among the external devices.

Abstract: A sound system is shown in the form of a speaker with or without a speaker box. A front member may be used with an elongate open portion with a top end and a bottom end and a speaker support with the speaker mounted to the speaker support. The speaker support may be coupled to the front member with a front of the speaker facing the front member and the front of the speaker substantially aligned with the bottom end of the elongate open portion. The bottom end of the elongate open portion may be positioned adjacent to a lower portion of the back of the torso of a user and the top end of the open portion is positioned higher near the back of the torso of the user. This combination may be used in a backpack, a chair or any other similar device.

Abstract: Method and System for avoiding howling disturbance especially on conferences, wherein the method comprising the steps of using a howling detector unit implemented inside a multipoint control unit to receive an audio stream input from a client, analyzing the audio input with the howling detector in order to verify if howling noise is present, using at least two of a skewness analysis, a flatness analysis, a crest analysis, a rolloff analysis and preventing the audio stream input to be forwarded as an output to an audio mixer, if howling noise is present.

Abstract: The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: A method to decode audio signals is provided that includes: receiving an input spatial audio signal; determining directions of arrival of directional audio sources represented in the received input spatial audio signal; determining one of an active input spatial audio signal component and a passive spatial audio signal input component, based upon the determined directions of arrival; determining the other of the active input spatial audio signal component and the passive input spatial audio signal component based upon the determined one of the active input spatial audio signal component and the passive input spatial audio signal component; decoding the active input spatial audio signal component to a first output format; and decoding the passive input spatial audio signal component to a second output format.

Abstract: Apparatus including a processor configured to: receive at least two microphone audio signals; determine spatial metadata associated with the at least two microphone audio signals; and synthesize adaptively a plurality of spherical harmonic audio signals based on at least one microphone audio signal and the spatial metadata in order to output a pre-determined order spatial audio signal format.

Abstract: An audio system is described that includes one or more speaker arrays that emit sound corresponding to one or more pieces of sound program content into associated zones within a listening area. Using parameters of the audio system (e.g., locations of the speaker arrays and the audio sources), the zones, the users, the pieces of sound program content, and the listening area, one or more beam pattern attributes may be generated. The beam pattern attributes define a set of beams that are used to generate audio beams for channels of sound program content to be played in each zone. The beam pattern attributes may be updated as changes are detected within the listening environment. By adapting to these changing conditions, the audio system is capable of reproducing sound that accurately represents each piece of sound program content in various zones.

Abstract: A method to encode audio signals is provided for use with an audio capture device that includes multiple microphones having a spatial arrangement on the device, a method to encode audio signals comprising: receiving multiple microphone signals corresponding to the multiple microphones; determining a number and directions of arrival of directional audio sources represented in the one or more microphone signals; determining one of an active microphone signal component and a passive microphone signal component, based upon the determined number and directions of arrival; determining the other of the active microphone signal component and the passive microphone signal component, based upon the determined one of the active input spatial audio signal component and the passive input spatial audio signal component; encoding the active microphone signal component; encoding the passive microphone signal component.

Abstract: The pilot communication system employs a transducer array of individual spaced apart speakers and/or microphones for deployment within the cockpit. A signal processing circuit interfaces with the transducer array and also with the aircraft avionics communication system, and selectively applies different time delays to the individual speakers and/or microphones to create in the array an acoustic beam having steerable coverage within the acoustic space of the cockpit. By adjusting the time delays the signal processing circuit directs and focuses sound from the speakers to the pilot and similarly focuses the microphones on the pilot's mouth. In this way the pilots can communicate with each other and with air traffic control without the need to wear headsets. The system also significantly reduces flight deck warnings from being introduced into the cabin environment.

Abstract: Various techniques for driving phased array systems are described, specifically intended for acoustic phased arrays with applications to mid-air haptics, parametric audio, acoustic levitation and acoustic imaging, including a system: 1) that is capable of mitigating the effect of the changes in the air to provide a consistent haptic experience; 2) that produces trap points in air; 3) that defines phased-array optimization in terms of vectors for the production of more consistent haptic effects; 4) that defines one or more control points or regions in space via a controlled acoustic field; 5) that uses a reduced representation method for the construction of acoustic basis functions; 6) that performs efficient evaluation of complex-valued functions for a large quantity of throughput; 7) that generates a Krylov sub-space of a matrix; and 8) that maximizes an objective described by different control points and/or regions to those used to create the acoustic basis functions.

Abstract: A sound reproducing apparatus capable of self-diagnostic. The apparatus includes a directional speaker emitting ultrasound waves to a target object, an information acquisition unit configured to acquire a sound from the target object and optionally an image of the target object, and a processor electrically connected with the directional speaker and the information acquisition unit. The processor drives the directional speaker to emit the ultrasound waves to the target object and diagnoses a failure of the directional speaker based on the sound acquired by the information acquisition unit. A self-diagnostic method for a sound reproducing apparatus having a directional speaker is also provided.

Abstract: The present disclosure is provided with: at least one audio output unit each including multiple speaker units, at least one of the speaker units in each audio output unit being arranged in orientation different from orientation or orientations of the other speaker units; and an audio signal rendering unit configured to perform rendering processing of generating audio signals to be output from each of the speaker units, based on input audio signals, wherein the audio signal rendering unit performs first rendering processing on a first audio signal included in the input audio signals and performs second rendering processing on a second audio signal included in the input audio signals, and the first rendering processing is rendering processing that enhances a localization effect more than the second rendering processing does.

Abstract: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: Techniques for using beam-formed acoustic notifications for pedestrian notification are described. Computing device(s) can receive sensor data associated with an object in an environment of a vehicle. The computing device(s) can determine first data for emitting a first beam of acoustic energy via speaker(s) of an acoustic array associated with the vehicle, and second data for emitting a second beam of acoustic energy via speakers of the acoustic array. The computing device(s) can cause the speaker(s) to emit the first beam in a direction of the object at a first time and the second beam in the direction of the object at a second time. Directions of propagation of the first beam and the second beam are offset so that the object can localize the source the acoustic notification, thereby localizing the vehicle in the environment.

Abstract: The present document describes a method for extracting J audio sources from I audio channels. The method includes updating a Wiener filter matrix based on a mixing matrix from a source matrix and based on a power matrix of the J audio sources. Furthermore, the method includes updating a cross-covariance matrix of the I audio channels and of the J audio sources and an auto-covariance matrix of the J audio sources, based on the updated Wiener filter matrix and based on an auto-covariance matrix of the I audio channels. In addition, the method includes updating the mixing matrix and the power matrix based on the updated cross-covariance matrix of the I audio channels and of the J audio sources, and/or based on the updated auto-covariance matrix of the J audio sources.

Abstract: The method and apparatus described herein make use of multiple sets of head related transfer functions (HRTFs) that have been synthesized or measured at various distances from a reference head, spanning from the near-field to the boundary of the far-field. Additional synthetic or measured transfer functions maybe used to extend to the interior of the head, i.e., for distances closer than near-field. In addition, the relative distance-related gains of each set of HRTFs are normalized to the far-field HRTF gains.

Abstract: The disclosure relates to an audio signal processing apparatus for processing an input audio signal, comprising a filter unit comprising a plurality of filters, each filter configured to filter the input audio signal to obtain a plurality of filtered audio signals, each filter designed according to an extended mode matching beamforming applied to a surface of a half revolution, the surface partially characterizing a loudspeaker enclosure shape, a plurality of scaling units, each scaling unit configured to scale the plurality of filtered audio signals using a plurality of gain coefficients to obtain a plurality of scaled filtered audio signals, and a plurality of adders, each adder configured to combine the plurality of scaled filtered audio signals, thereby providing an output audio signal for producing a sound field having a beam directivity pattern defined by the plurality of gain coefficients.

Abstract: A method to decode audio signals is provided that includes: receiving an input spatial audio signal; determining directions of arrival of directional audio sources represented in the received input spatial audio signal; determining one of an active input spatial audio signal component and a passive spatial audio signal input component, based upon the determined directions of arrival; determining the other of the active input spatial audio signal component and the passive input spatial audio signal component, based upon the determined one of the active input spatial audio signal component and the passive input spatial audio signal component; decoding the active input spatial audio signal component to a first output format; and decoding the passive input spatial audio signal component to a second output format.

Abstract: A method for generating loudspeaker signals associated with a target screen size is disclosed. The method includes receiving a bit stream containing encoded higher order ambisonics signals, the encoded higher order ambisonics signals describing a sound field associated with a production screen size. The method further includes decoding the encoded higher order ambisonics signals to obtain a first set of decoded higher order ambisonics signals representing dominant components of the sound field and a second set of decoded higher order ambisonics signals representing ambient components of the sound field. The method also includes combining the first set of decoded higher order ambisonics signals and the second set of decoded higher order ambisonics signals to produce a combined set of decoded higher order ambisonics signals.

Abstract: According to an example aspect of the present invention, there is provided a method comprising: calculating a head-related transfer function for a user, the calculation based at least in part on gathered data comprising at least video material and at least one supplemental image, the at least one supplemental image comprising an image of a transparent measurement device superimposed over at least one biological detail of an user, and transmitting said head-related transfer function.

Abstract: Embodiments are directed to upward-firing speakers that reflect sound off a ceiling to a listening location at a distance from a speaker. The reflected sound provides height cues to reproduce audio objects that have overhead audio components. A virtual height filter based on a directional hearing model is applied to the upward-firing driver signal to improve the perception of height for audio signals transmitted by the virtual height speaker to provide optimum reproduction of the overhead reflected sound. The upward firing driver is tilted at an inclination angle of approximately 20 degrees to the horizontal axis of the speaker and separate height and direct terminal connections are provided to interface to an adaptive audio rendering system.

Abstract: An audio system provides for crosstalk processing and crosstalk compensation processing of an audio signal. The crosstalk processing may include crosstalk cancellation processing or crosstalk simulation processing. A crosstalk processed signal is generated by applying the crosstalk processing to a side channel of the left and right channels, with a mid channel of the left and right channels bypassing the crosstalk processing. The crosstalk processed signal and the mid channel that bypasses crosstalk processing is used to generate a left output channel and a right output channel. In some embodiments, a crosstalk compensated signal is generated by applying crosstalk compensation processing to the side channel. The crosstalk compensated signal adjusts for spectral defects caused by the crosstalk processing. The crosstalk processing and crosstalk compensation processing may be applied in different orders.

Abstract: Audio signals are received. The audio signals include left and right surround channels. The audio signals are played back using far-field loudspeakers distributed around a space having a plurality of listener positions. The left and right surround channels are played back by a pair of far-field loudspeakers arranged at opposite sides of the space having the plurality of listener positions. An audio component coinciding with or approximating audio content common to the left and right surround channels is obtained. The audio component is played back using at least a pair of near-field transducers arranged at one of the listener positions. Associated systems, methods and computer program products are provided. Systems, methods and computer program products providing a bitstream comprising the audio signals and the audio component are also provided, as well as a computer-readable medium with data representing such audio content.

Abstract: A method for localization of sound in a speaker system comprises determining speaker locations of a plurality of speakers in a speaker system, determining a user location of a user within a room, and modifying audio signals to be transmitted to each of the plurality of speakers based on the user location in the room relative to a corresponding one of the speaker locations. An optimum modification of the audio signals for each of the plurality of speakers includes eliminating locational effects of the user location within the room.

Abstract: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

Abstract: A device includes a receiver and a decoder. The receiver is configured to receive bitstream parameters corresponding to at least an encoded mid signal. The decoder is configured to generate a synthesized mid signal based on the bitstream parameters. The decoder is also configured to generate a synthesized side signal selectively based on the bitstream parameters in response to determining whether the bitstream parameters correspond to an encoded side signal.

Abstract: Examples involve calibration of two synchronous playback groups. In an example implementation, a first playback device forms a first group with a second playback device. While formed into the first group, the first playback device plays back a first calibration sound contemporaneously with the second playback device. The first playback device forms a second group with a third playback device. While formed into the second group, the first playback device repeatedly plays back a second calibration sound contemporaneously with the third playback device. The first playback device receives data representing (i) first audio processing coefficients and (ii) second audio processing coefficients. The first playback devices applies the first audio processing coefficients to playback of audio content when formed into the first group and the second audio processing coefficients to playback of audio content when formed into the second group.

Abstract: An acoustic array system includes a sound field controller and an acoustic transducer array. The sound field controller provides first and second processed signals. The first processed signal is associated with a first acoustic radiation pattern and the second processed signal is associated with a second acoustic radiation pattern. The transducer array receives the first and second processed signals from the sound field controller and produces first and second driver signals for each of the transducers. The first driver signals are based upon the first processed signal and the second driver signals are based upon the second processed signal. The transducer array combines the first and second driver signals for each of the transducers to produce a plurality of combined driver signals, one for each of the transducers, and provides the combined driver signals to the transducers.

Abstract: A method of creating a multichannel audio signal by: determining an expected series of audio emission source locations around an expected listener location; determining a surface around the expected listener location, the surface including the expected series of audio emission source locations; mapping an audio object location into a surface energy component having a surface energy location and magnitude and an expected listener location energy component having an expected listeners location energy location and magnitude; panning an audio object signal for the surface energy component to surrounding expected audio emission sources to produce a first set of surface panned audio emission signals; panning the audio object signal for the expected listeners location energy location to surrounding expected audio emission sources to produce a second set of expected listeners location panned audio emission signals; combining the first and second set of panned audio signals to produce the multichannel audio signal.

Abstract: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

Abstract: Disclosed is an audio signal processing method. The audio signal processing method according to the present invention comprises the steps of: receiving a bit-stream including at least one of a channel signal and an object signal; receiving a user's environment information; decoding at least one of the channel signal and the object signal on the basis of the received bit-stream; generating the user's reproducing channel information on the basis of the user's received environment information; and generating a reproducing signal through a flexible renderer on the basis of at least one of the channel signal and the object signal and the user's reproducing channel information.

Abstract: An audio system includes a loudspeaker cabinet defining a longitudinal axis. Several loudspeaker transducers are distributed around the longitudinal axis. The audio system includes an audio rendering processor to cause the loudspeaker transducers to emit a sound field approximating a desired contour. The desired contour can be decomposed into a combination of several constituent modal beam components, and the audio rendering processor can render a truncated version of the decomposition to render an approximation of the desired contour. The desired contour can be one of a plurality of contours stored in a memory, or can be user defined. The cabinet includes a processor and a memory having instructions that, when executed by the processor, cause the audio system decompose a desired contour and to render a truncated version of the decomposition. Related principles are described by way of reference to method and apparatus examples.

Abstract: The present technology relates to a sound processing apparatus and a sound processing system for enabling more stable localization of a sound image. A virtual speaker is assumed to exist on the lower side among the sides of a tetragon having its corners formed with four speakers surrounding a target sound image position on a spherical plane. Three-dimensional VBAP is performed with respect to the virtual speaker and the two speakers located at the upper right and the upper left, to calculate gains of the two speakers at the upper right and the upper left and the virtual speaker, the gains being to be used for fixing a sound image at the target sound image position. Further, two-dimensional VBAP is performed with respect to the lower right and lower left speakers, to calculate gains of the lower right and lower left speakers, the gains being to be used for fixing a sound image at the position of the virtual speaker.

Abstract: In general, techniques are described for crossfading sets of spherical harmonic coefficients. An audio encoding device or audio decoding device comprising a memory and a processor may be configured to perform the techniques. The memory may be configured to store a first set of spherical harmonic coefficients (SHCs) and a second set of SHCs. The first set of SHCs describe a first sound field. The second set of SHCs describe a second sound field. The processor may be configured to crossfade between the first set of SHCs and a second set of SHCs to obtain a first set of crossfaded SHCs.

Abstract: Examples involve calibration of a synchronous playback group. In an example implementation, a first playback device forms a synchronous playback group with a second playback device. While part of the synchronous playback group, the first playback device receives a command to repeatedly emit a calibration sound according to a sequence for duration of time. Based on the command, the first playback device emits the calibration sound contemporaneously with emission of the calibration sound by the second playback device. Emission of the calibration sound by the first playback device and emission of the calibration sound by the second playback device are staggered such that each emitted calibration sound is delayed relative to a preceding emitted calibration sound. After the duration of time, the first playback device receives data indicating audio processing coefficients and applies the audio processing coefficients when playing back audio content in the synchronous playback group.

Abstract: Various methods and devices for vehicles are provided, wherein directional output of sound is used. In some embodiments, speech signals assigned to different speakers in a telephone conference are reproduced with different directional characteristics. In other embodiments, menu items of a menu are reproduced using different directions. In some embodiments, a loudspeaker arrangement may be provided at a dashboard or in a backrest of a seat of the vehicle.

Abstract: An example implementation may involve a network device detecting within a playback environment, via a microphone, an audio signal over a duration of time. After the duration of time, the implementation further involves the network device determining a frequency response of the playback environment based on the detected audio signal and a test tone. The test tone includes (i) a first component that includes noise at frequencies between a minimum of a calibration frequency range and a first threshold frequency, and (ii) a second component that sweeps through frequencies between a second threshold frequency and a maximum of the calibration frequency range.

Abstract: An apparatus for generating four or more audio output signals has a panning gain determiner and a signal processor. The panning gain determiner is configured to determine a proper subset from a set of five or more loudspeaker positions, so that the proper subset has four or more of the five or more loudspeaker positions. Moreover, the panning gain determiner is configured to determine the proper subset depending on a panning position and on the five or more loudspeaker positions, and to determine a panning gain for each of the four or more audio output signals by determining the panning gain depending on the panning position and on the four or more loudspeaker positions of the proper subset. The signal processor is configured to generate each of the four or more audio output signals depending on the panning gain for the audio output signal and on an audio input signal.

Abstract: Embodiments of systems and methods are described for estimating a position of a loudspeaker and notifying a listener if an abnormal condition is detected, such as an incorrect loudspeaker orientation or an obstruction in a path between the loudspeaker and a microphone array. For example, a front component of a multi-channel surround sound system may include the microphone array and a position estimation engine. The position estimation engine may estimate the distance between the loudspeaker and the microphone array. In addition, the position estimation engine may estimate an angle of the loudspeaker using a first technique. The position estimation engine may also estimate an angle of the loudspeaker using a second technique. The two angles can be processed to determine whether the abnormal condition exists. If the abnormal condition exists, a listener can be notified and be provided with suggestions for resolving the issue in a graphical user interface.

Abstract: Implementations disclosed herein relate to transmitting audio in a target space relative to an electronic device using a directional speaker and non-directional speaker. In one implementation, a directional speaker or non-directional speaker is selected to transmit audio based on a target space for the audio. The audio may then be transmitted from the selected speaker.

Abstract: An audio object encoder comprises a receiver (701) which receives N audio objects. A downmixer (703) downmixes the N audio objects to M audio channels, and a channel circuit (707) derives K audio channels from the M audio channels, K=1, 2 and K<M. A parameter circuit (709) generates audio object upmix parameters for at least part of each of the N audio objects relative to the K audio channels and an output circuit (705, 711) generates an output data stream comprising the audio object upmix parameters and the M audio channels. An audio object decoder receives the data stream and includes a channel circuit (805) deriving K audio channels from the M channel downmix; and an object decoder (807) for generating at least part of each of the N audio objects by upmixing the K audio channels based on the audio object upmix parameters. The invention may allow improved object encoding while maintaining backwards compatibility.