Abstract: An audio system and method of using the audio system to augment spatial audio rendition is described. The audio system can include a device to receive user inputs designating positions on an augmented reality view of a listening environment. Sound source icons can be presented in the augmented reality view at the designated positions. The sound source icons can visually represent sound sources at locations in the listening environment that correspond to, but are different than, the positions. One or more processors of the audio system can apply head-related transfer functions, which correspond to the locations in the listening environment, to audio input signals to generate binaural audio signals. The audio system can include a headset that uses the binaural audio signals to render spatialized audio localizing sounds to the locations in the listening environment. Other aspects are also described and claimed.

Abstract: A system can include a position sensor configured to output position data of a HWD. The system can include one or more processors configured to identify a first head angle of the HWD using the position sensor, generate an audio signal using the first head angle, identify a second head angle of the HWD using the position sensor, determine an angle error based at least on the first head angle and the second head angle, and apply at least one of a time difference or a level difference to the audio signal based at least on the angle error to adjust the audio signal. The system can include an audio output device configured to output the adjusted audio signal. By adjusting the audio signal using the angle error, the system can correct for long spatial update latencies and reduce the perceptual impact of such latencies for the user.

Abstract: An exemplary acoustics generation system accesses acoustic propagation data representative of characteristics affecting propagation of a virtual sound to an avatar within an extended reality world being experienced by a user associated with the avatar. Based on the acoustic propagation data, the acoustics generation system also generates a binaural audio signal representative of the virtual sound as experienced by the avatar when the propagation of the virtual sound to the avatar is simulated in accordance with the characteristics affecting the propagation. Additionally, the acoustics generation system prepares the binaural audio signal for presentation to the user as the user experiences the extended reality world by way of the avatar. Corresponding methods and systems are also disclosed.

Abstract: An audio system captures audio data of test sounds through a microphone of a headset worn by a user. The test sounds are played by an external speaker, and the audio data includes audio data captured for different orientations of the headset with respect to the external speaker. A set of head-related transfer function (HRTFs) is calculated based at least in part on the audio data of the test sounds at the different orientations of the headset. A portion of the set of HRTFs is discarded to create an intermediate set of HRTFs. The discarded portion corresponding to one or more distortion regions that are based in part on wearing the headset. One or more HRTFs are generated that correspond to the discarded portion using at least some of the intermediate set of HRTFs to create an individualized set of HRTFs for the user.

Abstract: The present technology relates to a sound processing device and a method that can present progress of sound reproduction. The sound processing device includes a control unit for controlling a sound output that aurally expresses progress of sound reproduction with respect to an entirety of the sound reproduction according to the reproduction of a sound. The present technology can be applied to a sound speech progress presentation UI system.

Abstract: An audio signal processing apparatus includes; a first correlation component separating unit configured to predict a first signal from a second signal in a predetermined period to generate a first correlation component signal and to separate the first non-correlation component signal from the first signal by using the first correlation component signal; a second correlation component separating unit configured to predict a second signal from the first signal in the predetermined period to generate a second correlation component signal and to separate the second non-correlation component signal from the second signal by using the second correlation component signal; a correlation component synthesizing unit configured to synthesize the first correlation component signal and the second correlation component signal to generate a synthesized correlation component signal; a first gain multiplying unit configured to multiply the synthesized correlation component signal by a gain to generate a correlation component

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: A speaker driving device includes a first calculation unit, a first driving signal generation unit and a third calculation unit. The first calculation unit outputs a first calculation signal obtained from a first input signal based on response characteristics according to a first parameter defining an equivalent circuit of a first speaker unit. The first driving signal generation unit generates a first driving signal based on a second driving signal and the first calculation signal. The first driving signal drives a first output speaker unit. The third calculation unit generates a third calculation signal from a second input signal based on response characteristics according to a third parameter defining an equivalent circuit of a third speaker unit. The second driving signal generation unit generates the second driving signal based on the first driving signal and the third calculation signal. The second driving signal drive a second output speaker unit.

Abstract: A filter determination system, a filter determination device, a filter determination method and a program with which a filter can be appropriately determined are provided. A processing device according to this embodiment includes a frequency response acquisition unit that acquires a frequency response of a filter for an audio signal, a smoothing unit that smoothes the frequency response and obtains a smoothed response, a candidate point determination unit that determines candidate split points based on a bottom position of the smoothed response, and a split point determination unit that determines one or more band split points from the candidate split points.

Abstract: An audio signal processing method. The audio signal processing method includes: acquiring a first stereo audio signal; splitting the first stereo audio signal into 5.1-channel audio signals; obtaining processed 5.1-channel audio signals by processing the 5.1-channel audio signals based on a speaker box parameter of a three-dimensional surround 5.1-channel virtual speaker box; and synthesizing the processed 5.1-channel audio signals into a second stereo audio signal.

Abstract: A display apparatus is provided. The display apparatus according to the disclosure includes a display, a plurality of speakers, a sensor, and a processor configured to control the display apparatus to: display an image and generate a sound corresponding to the image and output the sound through the plurality of speakers, and based on a user being sensed by the sensor, determine a location of the user, and based on the determined location of the user, change a first weight of a high band sound and a second weight of a low band sound of the sound, and output the changed sounds through the plurality of speakers.

Abstract: A distance measuring unit 331 measures a distance of a path of sound based on the number of map units between the map unit where a sound source is located and the map unit where a player character is located. An effect unit 332 attenuates the volume of a game sound based on the distance calculated by the distance measuring unit 331. Thereby, it is possible to attenuate the volume level of the sound effect depending on the distance where the sound is transmitted, even if the sound comes around an object obstructing sound.

Abstract: A portable audio device that includes a network card able to connect with a WLAN and a wireless modem to connect to a WWAN. The portable audio device communicates with a voices services platform and/or content provider via the network card and WLAN or the wireless modem and WWAN. If the portable audio device does not have access to the WLAN, the portable audio device may process and respond to voice queries by communicating with the voice services platform via the wireless modem and WWAN. The portable audio device also includes a battery that provides power for the various hardware and software components of the portable audio device to perform various functions, such as advanced voice functions. The portable audio device provides true portability and may be used in any environment, such as within a home or building environment or outside of the home or building environment.

Abstract: Systems and methods for performing noise reduction in a vehicle comprising a noise reduction system and a plurality of zones are herein provided. A noise reduction criterion for the vehicle is obtained. A first one of the plurality of zones for which noise reduction is to be performed is determined, based on the noise reduction criterion. The noise-reduction system is controlled to perform noise reduction in the first zone, wherein performing noise reduction in the first zone causes the noise reduction system to effect a resultant adjustment in noise level in at least a second one of the plurality of zones.

Abstract: An apparatus is disclosed, comprising at least one processor and at least one memory including computer program code, the at least one memory and the computer program code configured with the processor to cause the apparatus to perform the following steps. In a first step, the spatial position of a user in a real world space is detected. In another step, one or more audio signals are provided to the user representing audio from spatially-distributed audio sources in a virtual space. In another step, responsive to detecting movement of the user's spatial position from within a first zone to within a second zone within the real world space, the audio signals for selected ones of the audio sources are modified based on their spatial position in the virtual space.

Abstract: A hearing protection headset that can be worn by a user includes left and right earcups. The headset includes a radio communication system enabling at least one radio signal to be received and played through one or both of the earcups. Noise control circuitry in the headset is configurable by a user between at least three active modes of operation—the circuitry having a first active mode in which the headset provides automatic noise reduction, a second active mode in which the headset provides automatic noise cancellation, and a third active mode in which the headset provides both automatic noise reduction and automatic noise cancellation. A switch is manually operable by the user to configure the circuitry between the first active mode and the second active mode and the third active mode.

Abstract: An apparatus for outputting an audio signal includes: a channel processor configured to generate two or more channel signals from audio data; a signal processor configured to render the generated two or more channel signals; and a directional speaker configured to reproduced a rendered channel signal as an audible sound. The signal processor may include a frequency converter configured to generate a channel signal of a frequency domain by converting the generated two or more channel signals through frequency conversion, and a re-panner configured to change a channel gain of at least one of the generated channel signals by as much as an adjustment value for the channel gain, wherein the adjustment value is monotonically changed as a frequency of the channel signal of the frequency domain increases.

Abstract: The various implementations described herein include methods, devices, and systems for automatic audio equalization. In one aspect, a method is performed at an electronic device that includes speakers, microphones, processors and memory. The electronic device outputs audio user content from the speakers and automatically equalizes subsequent audio output of the device without user input. The automatic equalization includes: (1) obtaining audio content signals, including receiving outputted audio content at each microphone; (2) determining from the audio content signals phase differences between microphones; (3) obtaining a feature vector based on the phase differences; (4) obtaining a frequency correction from a correction database based on the obtained feature vector; and (5) applying the obtained frequency correction to the subsequent audio output.

Abstract: Disclosed herein are methods, systems, and computing devices for fitting bilateral hearing prostheses. An example method includes sending a signal to a first hearing prosthesis and a second hearing prosthesis. The signal causes the first hearing prosthesis to deliver a first stimulus to a body part in a left auditory pathway of a user. The signal also causes the second hearing prosthesis to deliver a second stimulus to a body part in a right auditory pathway of the user. The first stimulus and the second stimulus cause the user to perceive a sound and are delivered simultaneously. The method also includes receiving an indication of a perception of the sound by the user. The method further includes determining an adjustment to at least one of the first stimulus or the second stimulus based on the perception of the sound by the user.

Abstract: An audio signal processing system according to an aspect of the present invention includes an audio signal processing unit configured to select one rendering scheme among a plurality of rendering schemes, based on track information for indicating a reproduction position of an audio signal, and render the audio signal by using the one rendering scheme selected.

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: A method for processing a 3D audio effect and related products are provided. The method includes the following. In a VR scenario, eight first positions of eight monaural sound sources and eight pieces of monaural data corresponding to the eight monaural sound sources are obtained, where the eight monaural sound sources correspond to a target object, and each eight monaural sound source locates at a position with unique 3D coordinates. A second position of a target object in the VR scenario is obtained, where the second position has 3D coordinates, and the eight first positions and the second position are based on the same coordinate origin. A distance between each of the eight first positions and the second position is determined to obtain eight distances. The eight pieces of monaural data are adjusted according to the eight distances. The eight pieces of monaural data adjusted are synthesized to 7.1-channel data.

Abstract: An apparatus comprising means configured to: receive first audio content from a remote apparatus, the audio content comprising at least one audio stream associated with respective location information; present the audio stream as spatial audio to be perceived as originating from a respective perceived-direction based on the location information; determine a user input direction to identify one of the audio streams; transmit a message to the remote apparatus identifying the identified audio stream; receive second audio content comprising an enhanced version of the at least one identified audio stream; present the enhanced version of the identified audio stream as spatial audio, wherein the perceived-direction of the identified audio stream is offset from the second perceived-direction of the enhanced version of the identified audio stream.

Abstract: A computing device may be configured to receive a content asset and to determine whether the content asset comprises one or more triggers. The trigger may be a word, phrase, or passcode that alerts a voice activated device to the presence of a voice command and may serve as an instruction to the voice activated device to cause execution of the voice command. In response to determining that the content asset comprises one or more triggers, the computing device may be configured to insert one or more signal markers into the content asset at a location corresponding to the one or more triggers, and to cause transmission and/or presentation of the content asset with the one or more signal markers. The signal markers may cause a voice activated device to ignore a voice command in the content asset, despite the presence of one or more triggers.

Abstract: An audio system having a depth capturing device and a microphone array to respectively detect a point cloud and a local sound field, is described. The point cloud includes points corresponding to objects in a source space, and the local sound field includes sounds received in respective directions from the source space. The audio system includes one or more processors to generate a global sound field based on distances to the points and directions of the sounds. The global sound field includes virtual sound sources emitting respective sounds at respective points. A speaker can render the virtual sound sources to a user in a virtual space that corresponds to the source space.

Abstract: A personal audio assistant includes a first microphone for capturing audio, a logic circuit coupled to the first microphone for analyzing the captured audio, a communication module coupled to the logic circuit, and a user interaction element. The interaction element is configured to control at least one function or operation among invoking an operation of an electronic device communicatively coupled to the personal audio assistant via the communication module, invoking access to information on a network, invoking delivery of audio content, invoking delivery of text to speech, invoking speech to text processing of email, invoking delivery of text to speech of stock market information, invoking execution of a commercial transaction, invoking delivery of calendaring information, invoking delivery of medication reminders, or invoking access to location information using a GPS system. Other embodiments are disclosed.

Abstract: The present invention relates to a terminal and a method of operating the terminal. The terminal can execute at least one task. The terminal can execute at least one task, can cause the display to display a soft key and information related to the executed at least one task at a first region of the display and can cause the display to display at least one function indicator related to the information at a second region of the display.

Abstract: A conference system with transmitting and receiving sides. The transmitting side has a microphone array unit with microphone capsules, and a processing unit. The processing unit is configured to receive output signals of the microphone capsules and to execute audio beamforming based on the received output signals for acquiring sound coming from an audio source in a first direction. The processing unit has a direction-recognition unit that computes from the output signals of said microphone capsules a score for each of multiple search grid spatial positions and uses a search grid spatial position having a higher score to identify said first direction. The receiving side has an audio reproduction system that reproduces an audio signal detected by the microphone array with directional information of the first direction. The detected audio signal and the directional information regarding the first direction are transmitted from the transmitting side to the receiving side.

Abstract: An electronic apparatus includes a memory configured to store instructions; and a processor configured to execute the instructions to: separate an input audio signal into a plurality of channel signals; identify a gain corresponding to a feature difference between a first channel signal, from among the plurality of channel signals, and a second channel signal, from among the plurality of channel signals; and adjust relative ratios among a plurality of output signals according to the identified gain to generate an output audio signal in which a sound image is varied.

Abstract: A head mounted display (HMD) determines a distance between eyes of a user wearing the HMD, and head-related transfer functions (HRTFs) are selected based on the eye distance. The HMD displays a virtual image to the user and processes sound with the HRTFs into binaural sound that externally localizes at the virtual image. The HMD that determines the distance between the eyes of the user is also the HMD that displays the virtual image and processes the sound with the HRTFs.

Abstract: An audio processing apparatus has a setting processor that sets a size of a virtual sound source; and a signal processor that generates an audio signal by imparting to an audio signal a plurality of head-related transfer characteristics. The plurality of head-related transfer characteristics corresponds to respective points within a range that accords with the size set by the setting processor from among a plurality of points, with each point having a different position relative to a listening point.

Abstract: A system, computer-implemented method, and tangible computer-readable media for media playback. The method includes receiving a request from a user to play a media asset, detecting capabilities of the playback device, detecting playback environment acoustic characteristics, optimizing media playback settings for the playback device and the playback environment characteristics, preparing the media asset for playback on the playback device, and transferring the prepared media asset to the playback device for playback. The method can further transfer optimized playback settings to the playback device. Media asset preparation can be based at least in part on the optimized playback settings. A playback profile can store environment characteristics and playback device capabilities. The method continuously detects changes in the playback environment characteristics, and optimizes media playback settings when detected changes exceed a threshold.

Abstract: Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

Abstract: Systems and methods for audio communication are disclosed. The system includes a plurality of transducer units and plurality of three dimensional sensor modules (TDSMs) adapted to be located in a plurality of sites/spaces to which service should be provided by the system. The plurality of transducer units are capable of emitting/directing and focusing ultra-sonic signals to respective coverage zones in the sites, such that localized (confined) sound field can be formed at selected spatial position in the coverage zones by utilizing sound from ultrasound technique. The TDSMs are associated with respective sensing volumes in the sites and are operable to obtain sensory data indicative of the 3D arrangement of elements in a within the sites.

Abstract: An area reproduction method includes converting a sound pressure distribution at each frequency of a reproduced sound from a sound pressure distribution in a frequency domain into a sound pressure distribution in a spatial frequency domain. The reproduced sound is realized on a control line, including a reproduction line in which sound waves emitted from a speaker array including a plurality of speakers arranged intensify with each other, and a non-reproduction line in which the sound waves weaken with each other. The method includes determining a spatial frequency for use in adjustment of the reproduced sound, in the sound pressure distribution in the spatial frequency domain, based on a positional relationship between the speaker array and the control line, and adjusting a sound pressure of the reproduced sound, which is to be output by each of the plurality of speakers using the determined spatial frequency.

Abstract: An audio cancellation system includes a voice enabled computing system that is connected to an audio output device using a wired or wireless communication network. The voice enabled computing device can provide media content to a user and receive a voice command from the user. The connection between the voice enabled computing system and the audio output device introduces a time delay between the media content being generated at the voice enabled computing device and the media content being reproduced at the audio output device. The system operates to determine a calibration value adapted for the voice enabled computing system and the audio output device. The system uses the calibration value to filter the user's voice command from a recording of ambient sound including the media content, without requiring significant use of memory and computing resources.

Abstract: A three-dimensional audio playing method and playing apparatus are disclosed. The three-dimensional audio playing method according to the present invention comprises: a decoding step of decoding a received audio signal and outputting the decoded audio signal and metadata; a room impulse response (RIR) decoding step of decoding RIR data when the RIR data is included in the received audio signal; a head-related impulse response (HRIR) generation step of generating HRIR data by using user head information when the RIR data is included in the received audio signal; a binaural room impulse response (BRIR) synthesis step of generating BRIR data by synthesizing the decoded RIR data and modeled HRIR data; and a binaural rendering step of outputting a binaural rendered audio signal by applying the generated BRIR data to the decoded audio signal. In addition, the three-dimensional audio playing method and playing apparatus, according to the present invention, support a 3DoF environment and a 6DoF environment.

Abstract: Audio perception in local proximity to visual cues is provided. A device includes a video display, first row of audio transducers, and second row of audio transducers. The first and second rows can be vertically disposed above and below the video display. An audio transducer of the first row and an audio transducer of the second row form a column to produce, in concert, an audible signal. The perceived emanation of the audible signal is from a plane of the video display (e.g., a location of a visual cue) by weighing outputs of the audio transducers of the column. In certain embodiments, the audio transducers are spaced farther apart at a periphery for increased fidelity in a center portion of the plane and less fidelity at the periphery.

Abstract: An audio signal processing method, includes: acquiring 5.1-channel audio signals; acquiring head related transfer function (HRTF) data corresponding to each virtual speaker box in 5.1-channel virtual speaker boxes based on coordinates of the 5.1-channel virtual speaker boxes in a virtual environment; obtaining processed 5.1-channel audio signals by processing corresponding channel audio signals in the 5.1-channel audio signals based on the HRTF data corresponding to each virtual speaker box; and synthesizing the processed 5.1-channel audio signals into a stereo audio signal.

Abstract: Provided is a sound source separation device that includes a combining unit that combines a first sound source separation signal of a predetermined sound source, the first sound source separation signal being separated from a mixed sound signal by a first sound source separation system, with a second sound source separation signal of the sound source, the second sound source separation signal being separated from the mixed sound signal by a second sound source separation system that differs in separation performance from the first sound source separation system in predetermined units of time, and that outputs a sound source separation signal obtained by the combination.

Abstract: A system, method and computer product for training a neural network system. The method comprises applying an audio signal to the neural network system, the audio signal including a vocal component and a non-vocal component. The method also comprises comparing an output of the neural network system to a target signal, and adjusting at least one parameter of the neural network system to reduce a result of the comparing, for training the neural network system to estimate one of the vocal component and the non-vocal component. In one example embodiment, the system comprises a U-Net architecture. After training, the system can estimate vocal or instrumental components of an audio signal, depending on which type of component the system is trained to estimate.

Abstract: A handheld portable electronic device (HPED) defines a zone that extends from a floor and around a user wearing a wearable electronic device (WED). The HPED designates a location in the zone for where binaural sound originates to the user. The WED tracks a location of the HPED, and sound is processed to originate as the binaural sound in the zone. The WED displays a virtual image at the location of the binaural sound in the zone.

Abstract: A display apparatus includes: a display module including a display panel configured to display an image, a vibration plate on a rear surface of the display module, a connection member between the rear surface of the display module and the vibration plate, at least one vibration module on the vibration plate, and an enclosure between the display module and the vibration plate, the enclosure being spaced apart from the vibration module, the enclosure surrounding the vibration module.

Abstract: An audio metadata providing apparatus and method and a multichannel audio data playback apparatus and method to support a dynamic format conversion are provided. Dynamic format conversion information may include information about a plurality of format conversion schemes that are used to convert a first format set by a writer of multichannel audio data into a second format that is based on a playback environment of the multichannel audio data and that are set for each of playback periods of the multichannel audio data. The audio metadata providing apparatus may provide audio metadata including the dynamic format conversion information. The multichannel audio data playback apparatus may identify the dynamic format conversion information from the audio metadata, may convert the first format of the multichannel audio data into the second format based on the identified dynamic format conversion information, and may play back the multichannel audio data with the second format.

Abstract: An audio processing apparatus includes an interface, a control processor, an adjustment processor, channel modifiers, and a channel combiner. The interface is configured to receive audio channels including respective audio signals. The control processor is configured to generate a control signal from the audio signals. The adjustment processor is configured to calculate, based on the control signal, an adjusting parameter to an amplitude of at least one of the audio signals. The channel modifiers are configured to, using the adjusting parameter, adjust the audio signals in the respective audio channels. The channel combiner is configured to sum the audio channels after at least one channel has been adjusted, and output the summed audio channel to a user.

Abstract: A system generates an output audio signal for an object or virtual object using image data of a room to select a room impulse response from a database. A headset may include a depth camera assembly (DCA) and processing circuitry. The DCA generates depth image data of a room. The processing circuitry determines room parameters such as the dimensions of the room based on the depth image data. A room impulse response for the room is determined based on referencing a database of room impulse responses using the room parameters. An output audio signal is generated by convolving a source audio signal of an object with the room impulse response.

Abstract: An electronic device according to various embodiments of the present invention may comprise: a gyro sensor for sensing rotation of an electronic device; a memory for storing a plurality of head related transfer functions corresponding to a plurality of directions; and a processor configured to generate a stereo signal by applying, to an audio signal, a first head related transfer function corresponding to a first direction, generate a stereo signal by applying, to the audio signal, a second head related transfer function corresponding to a third direction symmetric to a second direction with reference to the first direction when the electronic device rotates from the first direction to the second direction, and generate a stereo signal by applying the first head related transfer function to the audio signal when a pre-designated time elapses after the electronic device rotates from the first direction to the second direction.

Abstract: A personal audio assistant includes a first microphone for capturing audio, a logic circuit coupled to the first microphone, a communication module coupled to the logic circuit, a memory storage unit communicatively coupled to the logic circuit, and a user interaction element. The interaction element is configured to perform at least one among accessing selection of audio content via an audio input received at the first microphone or determining a user preference of audio content by tracking genre of content selected, artist of content selected, and at least one among the day of the week or time of day. Other embodiments are disclosed.

Abstract: A communication system, a communication device and related method, the communication device comprising a processor, a source interface comprising a first source interface for a first external source, and an output interface, wherein the processor is configured to obtain, e.g. receive, a first input signal via the first source interface, obtain a first complementing signal, combine the first input signal and the first complementing signal to a first combined signal, apply a first filter function to the first combined signal for provision of a first left output signal and a first right output signal, the first filter function comprising one or more head related transfer functions, and output a left output signal and a right output signal via the output interface, wherein the left output signal is based on the first left output signal and the right output signal is based on the first right output signal.

Abstract: A first microphone on a first earpiece of a personal audio delivery device measures a first head related transfer function (HRTF) associated with a sound source. A second microphone on an earpiece of a personal audio delivery device measures a second HRTF associated with the sound source. An interaural time difference (ITD) is determined based on the first HRTF and second HRTF. A determination is made that the sound source is located in a first region based on the interaural time difference. A determination is made that the sound source is located in a second region within the first region based on the ITD of the first HRTF and the second HRTF and an ITD associated with third HRTFs for the second region. A location of the sound source is determined within the second region, which in some examples, is improved with a Kalman filter model.