Abstract: Techniques for utilizing three-dimensional (3D) sound for passenger notification are described herein. Computing device(s) onboard a vehicle can determine an occurrence of an event associated with a passenger of the vehicle or the vehicle, and can determine a 3D sound associated with the event. Then, the computing device(s) can send a signal associated with the 3D sound to a speaker system inside of the vehicle and, responsive to receiving the signal, one or more speakers of the speaker system can output the 3D sound such that a sound associated with the 3D sound is perceived to be localized relative to the passenger in the vehicle.

Abstract: An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Abstract: A method and apparatus for outputting an audio signal corresponding to a user position are disclosed. The method includes receiving an audio signal and providing a decoding audio signal and decoded metadata, checking whether a user position is changed in an arbitrary space using user position information including a user position change indicator and user position change offset, when the user position is changed, providing modified metadata obtained by correcting the decoded metadata based on the user position change offset, and rendering the decoded audio signal using the modified metadata. Accordingly, it is possible to provide an audio sound image that is changed in response to change in user position in an arbitrary space, thereby providing more realistic audio output.

Abstract: Techniques of performing binaural rendering involve combining left and right HRTFs from speakers symmetrically placed with respect to a forward axis of a listener's head. Because of the symmetry of the listening geometry, a decoded loudspeaker signal resulting from each speaker may be decomposed into symmetric and antisymmetric components according to whether a spherical harmonic associated with a respective ambisonic channel is symmetric or antisymmetric about the forward axis. The result involves rendering the left and right headphone speakers using the same sum and difference HRTFs rather than a different pair of HRFTs for each headphone speaker.

Abstract: Apparatus for mixing at least two audio signals, at least one audio signal associated with at least one parameter, and at least one second audio signal further associated with at least one second parameter, wherein the at least one audio signal and the at least one second audio signal are associated with a sound scene and wherein the at least one audio signal represent spatial audio capture microphone channels and the at least one second audio signal represents an external audio channel separate from the spatial audio capture microphone channels, the apparatus comprising: a processor configured to generate a combined parameter output based on the at least one second parameter and the at least one parameter; and a mixer configured to generate a combined audio signal with a same number or fewer number of channels as the at least one audio signal based on the at least one audio signal and the at least one second audio signal, wherein the combined audio signal is associated with the combined parameter.

Abstract: A method includes, estimating directional information based on multiple input channel signals representing at least one arriving sound from a sound source captured by respective multiple microphones that have respective known locations relative to each other, wherein said estimating comprises finding a time delay that removes a time difference between said first and second input channel signals; deriving a mid-signal and a side signal on a basis of a first input channel signal, a second input channel signal and said estimated directional information; and generating an output signal comprising a plurality of output channels using said mid-signal, said side signal and said estimated directional information such that the output signal retains a spatial representation of the captured at least one arriving sound. Apparatus and program products are also disclosed.

Abstract: A method, apparatus and computer program product are provided to cause AN audio source to be modified in a manner consistent with the corresponding video images once the user and/or a display upon which the images are rendered has been tilted. In regards to a method, an initial virtual position of an audio source is determined. The method also includes determining a tilt angle that defines an angle that an apparatus embodying a display for rendering images has been tilted relative to a reference orientation of the apparatus with respect to a user of the apparatus. The method may also include modifying a virtual position of an audio source based upon the tilt angle and an initial virtual position of the audio source. The method may also include causing the audio source to be rendered in accordance with the virtual position as modified.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which is represented by one or more audio signals. The encoder generates a bit stream which comprises downmix signals and side information which includes individual matrix elements of a reconstruction matrix which enables reconstruction of the one or more audio signals in the decoder.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: An audio signal processing apparatus, comprising a memory configured to store a set of pairs of predefined left ear and right ear transfer functions, which are predefined for a plurality of reference positions relative to the listener, a processor configured to determine a pair of left ear and right ear transfer functions on the basis of the set of predefined pairs of predefined left ear and right ear transfer functions for the azimuth angle and the elevation angle of the virtual target position and an adjustment filter configured to filter the input audio signal on the basis of the determined pair of left ear and right ear transfer functions and an adjustment function configured to adjust a delay between the left ear transfer function and the right ear transfer function to obtain a left ear output audio signal and a right ear output audio signal.

Abstract: For generating 3D audio content from a two-channel stereo signal, the stereo signal (x(t)) is partitioned into overlapping sample blocks and is transformed into time-frequency domain. From the stereo signal directional and ambient signal components are separated, wherein the estimated directions of the directional components are changed by a predetermined factor, wherein, if changes are within a predetermined interval, they are combined in order to form a directional center channel object signal. For the other directions an encoding to Higher Order Ambisonics (HOA) is performed. Additional ambient signal channels are generated by de-correlation and rating by gain factors, followed by encoding to HOA. The directional HOA signals and the ambient HOA signals are combined, and the combined HOA signal and the center channel object signals are transformed to time domain.

Abstract: A residual scaling unit is configured to determine a scaling factor for a residual channel based on an inter-channel mismatch value. The inter-channel mismatch value is indicative of a temporal misalignment between a reference channel and a target channel. The residual scaling unit is further configured to scale (e.g., attenuate) the residual channel by the scaling factor to generate a scaled residual channel. A residual channel encoder is configured to encode the scaled residual channel as part of a bitstream.

Abstract: An audio signal associated with a navigation prompt is processed, based on the localization and distance information supplied by a navigation system, to generate an audio signal that, when played by an audio system in the vehicle, emulates the localization and distance information.

Abstract: A new binaural hearing aid system is provided with a hearing aid in which signals that are received from external devices, such as a spouse microphone, a media player, a hearing loop system, a teleconference system, a radio, a TV, a telephone, a device with an alarm, etc., are filtered with binaural filters in such a way that a user perceives the signals to be emitted by respective sound sources positioned in different spatial positions in the sound environment of the user, whereby improved spatial separation of the different sound sources is facilitated.

Abstract: A system for enabling a shared three-dimensional (“3D”) audio bed available to multiple software applications is provided. The system manages bed metadata defining a number of speaker objects of a 3D audio bed. The bed metadata also associates each speaker object with a location, which in some configurations, is defined by a three-dimensional coordinate system. The bed metadata is communicated to a plurality of applications. The applications can then generate custom 3D audio data that associates individual audio streams with individual speaker objects of the 3D audio bed. The applications can then communicate the custom 3D audio data to a 3D audio bed engine, which causes the processing and rendering of the custom 3D audio data to an output device utilizing a selected spatialization technology. Aspects of the 3D bed can be altered when the spatialization technology or the output device changes.

Abstract: A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: An audio enhancement device including an audio-calculating module, a ratio-calculating module, a minimum-tracking module, a weighting-calculating module and a mixing module is provided. The audio-calculating module calculates a mid signal and a side signal according to a sum and a difference of an input first channel signal and an input second channel signal. The ratio-calculating module calculates a side-mid ratio of the side signal relative to the mid signal. The minimum-tracking module tracks a side-mid ratio minimum. The weighting-generating module determines a first and a second weighting values according to the side-mid ratio minimum. The mixing module weights the mid signal and the side signal based on the first and the second weighting values respectively and adjusts the input first channel signal and the input second channel signal accordingly to generate an enhanced first channel signal and an enhanced second channel signal.

Abstract: Methods, systems, and apparatuses are disclosed for generating a spatial audio format. An input audio source may include one or more individual channels. The one or more individual channels may be designated to be played by a corresponding one or more speakers. The one or more individual channels of the audio source may be separated. The one or more individual tracks may be input into a modeling space representing a multi-dimensional space. The modeling space may include a plurality of emitters at various locations in a vector space. Each of the one or more individual channels may be panned to one or more of the plurality of emitters. The panning may be based on a normalized proximity of the one or more individual channels in the modeling space to the plurality of emitters. The one or more of the plurality of emitters may be encoded into a single multichannel file.

Abstract: An ear shape analyzer includes: a sample ear analyzer configured to generate, for each of N sample ears, an ear shape data set that represents a difference between a point group representative of a three-dimensional shape of a reference ear and a point group representative of a three-dimensional shape of one of the N sample ears; an averaging calculator configured to generate averaged shape data by averaging N ear shape data sets generated by the sample ear analyzer; an ear shape identifier configured to identify an average ear shape of the N sample ears by translating coordinates of respective points of the point group representing the three-dimensional shape of the reference ear, by using the averaged shape data.

Abstract: Provided are methods and apparatus for enhancing a signal-to-noise ratio. In an example, provided is an apparatus configured to modify audio to better match the way the human brain processes audio by modifying the audio to a form which takes advantage of human echolocation capabilities. When humans listen to audio, they subconsciously listen for an echo and thus subconsciously focus on listening to, and for, meaningful information in audio. The focus causes humans to ignore noise in the audio, which results in enhancing a signal-to-noise ratio. In an example, the provided apparatus compensates for shortcomings of a device to which the apparatus is coupled by adjusting a respective amplitude of at least one constituent audio frequency of an output digital audio stream of the apparatus.

Abstract: The present technology relates to a signal processing device, a signal processing method, and a program, which are capable of reproducing an acoustic field more appropriately in accordance with content. A decoding unit decodes a multiplexed signal, and obtains a multichannel sound collection signal obtained by performing sound collection through a linear microphone array and spatial correction information for selecting a spatial correction scheme for correcting a spatial transfer characteristic. A spatial correction scheme selecting unit selects the spatial correction scheme on the basis of the spatial correction information, and a spatial transfer characteristic matrix generating unit outputs a spatial transfer characteristic matrix indicated by a selection result of the spatial correction scheme. A drive signal generating unit generates a speaker drive signal of a spatial frequency domain on the basis of the multichannel sound collection signal and the spatial transfer characteristic matrix.

Abstract: A propagation delay tune correction apparatus comprising a means for generating a frequency spectrum signal by performing short-term Fourier transform on an audio signal; a means for setting a propagation delay time for each of a plurality of predetermined frequency bands; a means for calculating a phase control amount for each of the plurality of predetermined frequency bands on a basis of the propagation delay time set for each of the plurality of predetermined frequency bands; a means for generating a phase control signal by smoothing the calculated phase control amount for each of the plurality of predetermined frequency bands; a means for controlling a phase of the frequency spectrum signal for each of the plurality of predetermined frequency bands on a basis of the generated phase control signal; and a means for generating an audio signal on which a propagation delay correction is performed by performing inverse short-term Fourier transform on the frequency spectrum signal of which the phase is controll

Abstract: An audio apparatus includes an analog module that receives a digital left (L) channel signal and a digital right (R) channel signal and outputs first and second analog L signals and first and second analog R signals, a first output port that includes first to fifth conductors, through which the first and second analog L signals, the first and second analog R signals, and a ground voltage are provided, and a second output port that includes sixth to ninth conductors, through which the first and second analog L signals and the first and second analog R signals are provided. When an audio jack including first to third terminals is inserted into the first output port, the first and second conductors are connected to the first terminal, the third and fourth conductors are connected to the second terminal, and the fifth conductor is connected to the third terminal.

Abstract: Example embodiments disclosed herein relates to upmixing of audio signals. A method of upmixing an audio signal is described. The method includes decomposing the audio signal into a diffuse signal and a direct signal, generating an audio bed at least in part based on the diffuse signal, the audio bed including a height channel, extracting an audio object from the direct signal, estimating metadata of the audio object, the metadata including height information of the audio object; and rendering the audio bed and the audio object as an upmixed audio signal, wherein the audio bed is rendered to a predefined position and the audio object is rendered according to the metadata. Corresponding system and computer program product are described as well.

Abstract: A stereo parameter conditioner performs a conditioning operation on a first value of a stereo parameter and a second value of the stereo parameter to generate a conditioned value of the stereo parameter. The first value is associated with a first frequency range, and the second value is associated with a second frequency range. The conditioned value is associated with a particular frequency range that is a subset of the first frequency range or a subset of the second frequency range.

Abstract: Exemplary embodiments provide encoding and decoding methods, and associated encoders and decoders, for encoding and decoding of an audio scene which at least comprises one or more audio objects (106a). The encoder (108, 110) generates a bit stream (116) which comprises downmix signals (112) and side information which includes individual matrix elements (114) of a reconstruction matrix which enables reconstruction of the one or more audio objects (106a) in the decoder (120).

Abstract: Disclosed is an audio signal processing device for performing binaural rendering on an input audio signal. The audio signal processing device includes a reception unit configured to receive the input audio signal, a binaural renderer configured to generate a 2-channel audio by performing binaural rendering on the input audio signal, and an output unit configured to output the 2-channel audio. The binaural renderer performs binaural rendering on the input audio signal based on a distance from a listener to a sound source corresponding to the input audio signal and a size of an object simulated by the sound source.

Abstract: The present invention relates to a neckband-type wireless sound transducer, and more particularly, to a neckband-type wireless sound transducer that prevents sound from being transferred to the surroundings of the wearer. The neckband-type wireless sound transducer according to the present invention, includes: a main body part configured to be seated on a human body; a main speaker part mounted on the main body part and configured to emit sound to the inside of the main body part; an auxiliary speaker part mounted on the main body part and configured to emit sound for leakage reduction to offset the sound emitted by the main speaker part and leaked to the outside of the main body part; and a controller configured to apply an electric signal for sound emission to the main speaker part and an electric signal for leakage reduction to the auxiliary speaker part.

Abstract: An audio navigation device comprising an input means for inputting two or more audio pieces into the navigation device; a spatialization means for allocating a position in the form of a unique spatial co-ordinate to each audio piece and arranging the audio pieces in a multi-dimensional arrangement; a generating means for generating a binaural audio output (3) for each audio piece, wherein the audio output (3) simulates sounds that would be made by one or more physical sources located at the given position of each audio piece; an output means for simultaneously outputting multiple audio pieces as binaural audio output (3) to a user (5); a navigation means (1) for enabling a user (5) to navigate around the audio output (3) in the multi-dimensional arrangement; a selection means (A) for allowing a user (5) to select a single audio output (3).

Abstract: Provided are a method and device for improving sound quality of stereo sound and a terminal. The method includes: an original left channel signal and an original right channel signal are acquired; phases, frequency spectrums and amplitudes of the original left channel signal and original right channel signal are acquired; a left calibrated signal is acquired according to the phase, frequency spectrum and amplitude of the original left channel signal, and a right calibrated signal is acquired according to the phase, frequency spectrum and amplitude of the original right channel signal; the left calibrated signal and the original right channel signal are superposed to generate a final right channel output signal; the right calibrated signal and the original left channel signal are superposed to generate a final left channel output signal; the final right channel output signal and the final left channel output signal are combined to form a PCM signal.

Abstract: An enhanced Stereo Dimensional Array loudspeaker system 250 preferably including a mirror image pair of loudspeaker enclosures 280L, 280R configurable by a user or installer as a left-channel loudspeaker and a right channel loudspeaker each having a driver array aiming configuration with first and second angled baffle facets carrying main and effects drivers on separate facets and a Head Shadow filter signal processing system and method for driving the main and effects drivers to achieve a psycho-acoustically expanded image breadth by Head Shadow filter compensated inter-aural crosstalk cancellation.

Abstract: Embodiments are described for an adaptive audio system that processes audio data comprising a number of independent monophonic audio streams. One or more of the streams has associated with it metadata that specifies whether the stream is a channel-based or object-based stream. Channel-based streams have rendering information encoded by means of channel name; and the object-based streams have location information encoded through location expressions encoded in the associated metadata. A codec packages the independent audio streams into a single serial bitstream that contains all of the audio data. This configuration allows for the sound to be rendered according to an allocentric frame of reference, in which the rendering location of a sound is based on the characteristics of the playback environment (e.g., room size, shape, etc.) to correspond to the mixer's intent.

Abstract: A spatial audio processor for providing spatial parameters based on an acoustic input signal has a signal characteristics determiner and a controllable parameter estimator. The signal characteristics determiner is configured to determine a signal characteristic of the acoustic input signal. The controllable parameter estimator for calculating the spatial parameters for the acoustic input signal in accordance with a variable spatial parameter calculation rule is configured to modify the variable spatial parameter calculation rule in accordance with the determined signal characteristic.

Abstract: A method of producing a spatialized stereo audio file from an original stereo audio file comprises creating a data base of impulse responses realized in at least one physical space divided into left, right, front, back, up and down sides relative to a sound acquisition position, with at least one pair of acquisition microphones placed at the sound acquisition position, with at least two pairs of source loudspeakers placed at sound source positions; the sound acquisition position is situated at the left-right median plane of the physical space, the sound source positions are distributed symmetrically by pairs relative to the sound acquisition position, the data base of impulse responses comprising at least one left/right impulse response pair, the left and right impulse responses being obtained by a deconvolution of the direct acquired signal from all the source loudspeakers distributed at the respective left and right side of the physical space.

Abstract: An audio system provides for spatial enhancement of an audio signal including a left input channel and a right input channel. The system may include a spatial frequency band divider, a spatial frequency band processor, and a spatial frequency band combiner. The spatial frequency band divider processes the left input channel and the right input channel into a spatial component and a nonspatial component. The spatial frequency band processor applies subband gains to subbands of the spatial component to generate an enhanced spatial component, and applies subband gains to subbands of the nonspatial component to generate an enhanced nonspatial component. The spatial frequency band combiner combines the enhanced spatial component and the enhanced nonspatial component into a left output channel and a right output channel. In some embodiments, the spatial component and nonspatial component are separated into spatial subband components and nonspatial subband components for the processing.

Abstract: A method, apparatus and computer program product are provided to cause AN audio source to be modified in a manner consistent with the corresponding video images once the user and/or a display upon which the images are rendered has been tilted. In regards to a method, an initial virtual position of an audio source is determined. The method also includes determining a tilt angle that defines an angle that an apparatus embodying a display for rendering images has been tilted relative to a reference orientation of the apparatus with respect to a user of the apparatus. The method may also include modifying a virtual position of an audio source based upon the tilt angle and an initial virtual position of the audio source. The method may also include causing the audio source to be rendered in accordance with the virtual position as modified.

Abstract: A transition between a stereophonic presentation and a monophonic presentation of a stereophonic input signal that includes a left channel signal and a right channel signal extracts content that is present at similar levels but not in-phase between the left and right channel signals to produce at least one of a left enhancement signal and a right enhancement signal. The left channel signal, the right channel signal, and only one of the left and right enhancement signals are combined to produce a monophonic signal for the monophonic presentation. Cross-fading between the left channel signal and the monophonic signal and between the right channel signal and the monophonic signal may be used to transition between the stereophonic presentation and the monophonic presentation. The stereophonic input signal may be up-mixed to produce enhancement signal. A similar transition between a multichannel presentation and a monophonic presentation of a multichannel signal is described.

Abstract: The invention discloses rendering sound field signals, such as Higher-Order Ambisonics (HOA), for arbitrary loudspeaker setups, where the rendering results in highly improved localization properties and is energy preserving. This is obtained by a new type of decode matrix for sound field data, and a new way to obtain the decode matrix.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: An equalizing device has: a low-frequency zero-point circuit having a zero point in a low-frequency band of a before-equalization frequency characteristic of a communication medium; a high-frequency zero-point circuit having a zero point in a high-frequency band of the before-equalization frequency characteristic of the communication medium; and an intermediate-frequency zero-point circuit having a zero point in an intermediate-frequency band present between the low-frequency band and the high-frequency band, wherein an inclination of a waveform of the before-equalization frequency characteristic of the communication medium changes in the intermediate-frequency band; wherein the equalizing device equalizes the signal transmitted through the communication medium so as to restrain an amount of change in an inclination of a waveform of the after-equalization frequency characteristic.

Abstract: An instant-on nothing-to-download “soundscaping” device that provides natural, atonal sounds such as the rolling surf of the ocean, running streams, gurgling brooks, rain, thunder, wind, crowd sounds, et al is provided.

Abstract: A method of estimating an individualized head-related transfer function (HRTF) and an individualized interaural time difference function (ITDF) of a particular person, comprising the steps of: a) obtaining a plurality of data sets (Li, Ri, Oi), each comprising a left and a right audio sample (Li, Ri) from a pair of in-ear microphones, and orientation information (Oi) from an orientation unit, measured in a test-arrangement whereby an acoustic test signal is rendered via a loudspeaker; b) storing the data sets in a memory; c) estimating the directions of the loudspeaker relative to the person based on the orientation data and the audio data; d) estimating the ITDF based on the data sets and on the estimated relative position/orientation; e) estimating the HRTF, based on the data sets and based on the estimated relative position/orientation.

Abstract: Methods and apparatus improve a user experience during telephone calls or other forms of communication in which a listener localizes electronically generated binaural sounds. The sound is convolved or processed to a location that is behind or near a source of the sound so that the listener perceives the location of the sound as originating from the source of the sound.

Abstract: A combo-plug detecting circuit for use in an audio CODEC is provided. The combo-plug detecting circuit is used to determine the contact configuration of the audio plug, to ensure that the audio plug belonging to the differential structure can be compatible with the audio CODEC.

Abstract: Certain aspects of the technology disclosed herein include generating a virtual sound field based on data from an ambisonic recording device. The ambisonic device records sound of a surrounding environment using at least four microphones having a tetrahedral orientation. An omnidirectional microphone having an audio-isolated portion can be used to isolate sound from a particular direction. Sound received from the plurality of microphones can be used to generate a virtual sound field. The virtual sound field include a dataset indicating a pressure signal and a plurality of velocity vectors. The ambisonic recording device can include a wide angle camera and generate wide angle video corresponding to the virtual sound field.

Abstract: Audio systems and methods are provided to reduce echo content in an audio signal. The systems and methods receive an audio signal and sound stage rendering parameter(s), and select a set of filter coefficients to filter the audio signal to provide an estimated echo signal. The estimated echo signal is subtracted from a microphone signal to provide an output signal with reduced echo content. The set of filter coefficients are selected based upon the sound stage rendering parameter(s) from among a plurality of stored sets of filter coefficients.

Abstract: The disclosure relates to an audio signal processing apparatus for filtering a left channel input audio signal (L) and a right channel input audio signal (R), a left channel output audio signal (X1) and a right channel output audio signal (X2) to be transmitted over acoustic propagation paths to a listener, wherein transfer functions of the acoustic propagation paths are defined by an acoustic transfer function matrix. The audio signal processing apparatus comprises a decomposer, a first cross-talk reducer, a second cross-talk reducer, and a combiner. The first cross-talk reducer is configured to reduce a cross-talk within a first predetermined frequency band upon the basis of the acoustic transfer function matrix. The second cross-talk reducer is configured to reduce a cross-talk within a second predetermined frequency band upon the basis of the acoustic transfer function matrix.

Abstract: A binaural decoder for an MPEG surround stream, which decodes an MPEG surround stream into a stereo 3D signal, and a decoding method thereof. The method includes dividing a compressed audio stream and head related transfer function (HRTF) data into subbands, selecting predetermined subbands of the HRTF data divided into subbands and filtering the HRTF data to obtain the selected subbands, decoding the audio stream divided into subbands into a stream of multi-channel audio data with respect to subbands according to spatial additional information, and binaural-synthesizing the HRTF data of the selected subbands with the multi-channel audio data of corresponding subbands.