Abstract: A device for processing audio signals includes an interchannel phase difference (IPD) mode selector and an IPD estimator. The IPD mode selector is configured to select an IPD mode from among at least a first IPD mode and a second IPD mode based on at least an interchannel temporal mismatch value indicative of a temporal misalignment between a first audio signal and a second audio signal. The IPD estimator is configured to determine IPD values based on the first audio signal and the second audio signal, the IPD values represented using a first number of bits responsive to selection of the first IPD mode or represented using a second number of bits responsive to selection of the second IPD mode.

Abstract: A voice extraction device according to the present invention includes a formation unit, an acquisition unit, an emphasis unit, a generation unit, and a selection unit. The formation unit forms directivity through beam-forming processing for each microphone in a microphone array including a plurality of microphones that form a plurality of channels. The acquisition unit acquires an observation signal that is a signal of voice received by each of the channels. The emphasis unit generates an emphasized signal by emphasizing the observation signal in accordance with the directivity formed by the formation unit. The generation unit generates, for each channel, frequency distribution of amplitude of the emphasized signal generated by the emphasis unit. The selection unit selects a channel corresponding to a voice signal used for voice recognition from among the channels based on the frequency distribution corresponding to the respective channels generated by the generation unit.

Abstract: A method for decoding an encoded audio bitstream in an audio processing system is disclosed. The method includes extracting from the encoded audio bitstream a first waveform-coded signal comprising spectral coefficients corresponding to frequencies up to a first cross-over frequency for a time frame and performing parametric decoding at a second cross-over frequency for the time frame to generate a reconstructed signal. The second cross-over frequency is above the first cross-over frequency and the parametric decoding uses reconstruction parameters derived from the encoded audio bitstream to generate the reconstructed signal. The method also includes extracting from the encoded audio bitstream a second waveform-coded signal comprising spectral coefficients corresponding to a subset of frequencies above the first cross-over frequency for the time frame and interleaving the second waveform-coded signal with the reconstructed signal to produce an interleaved signal for the time frame.

Abstract: An apparatus includes a receiver and an up-mixer. The receiver is configured to receive a bitstream that includes an encoded mid signal and encoded stereo parameter information. The encoded stereo parameter information represents a first value of a stereo parameter and a second value of the stereo parameter. The first value is associated with a first frequency range. The second value is associated with a second frequency range that is distinct from the first frequency range. The up-mixer is configured to perform an up-mix operation on a frequency-domain decoded mid signal generated from the encoded mid signal. A particular value based on the first value and the second value is applied to the frequency-domain decoded mid signal during the up-mix operation.

Abstract: An apparatus for spatial audio signal encoding, the apparatus comprising at least one processor and at least one memory including a computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to: determine, for two or more audio signals, at least one spatial audio parameter for providing spatial audio reproduction, the at least one spatial audio parameter comprising a direction parameter with an elevation and an azimuth component; define a spherical grid generated by covering a sphere with smaller spheres, the smaller spheres arranged in circles of spheres wherein a first circle of spheres comprises one of the smaller spheres located with a centre at an elevation of 90 degrees relative to a reference direction of the sphere; and convert the elevation and azimuth component of the direction parameter to an index value based on the defined spherical grid.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: An encoder and an encoding method for a multi-channel signal, and a decoder and a decoding method for a multi-channel signal are disclosed. A multi-channel signal may be efficiently processed by consecutive downmixing or upmixing.

Abstract: An encoder and an encoding method for a multi-channel signal, and a decoder and a decoding method for a multi-channel signal are disclosed. A multi-channel signal may be efficiently processed by consecutive downmixing or upmixing.

Abstract: The application relates to audio encoder and decoder systems. An embodiment of the encoder system comprises a downmix stage for generating a downmix signal and a residual signal based on a stereo signal. In addition, the encoder system comprises a parameter determining stage for determining parametric stereo parameters such as an inter-channel intensity difference and an inter-channel cross-correlation. Preferably, the parametric stereo parameters are time- and frequency-variant. Moreover, the encoder system comprises a transform stage. The transform stage generates a pseudo left/right stereo signal by performing a transform based on the downmix signal and the residual signal. The pseudo stereo signal is processed by a perceptual stereo encoder. For stereo encoding, left/right encoding or mid/side encoding is selectable. Preferably, the selection between left/right stereo encoding and mid/side stereo encoding is time- and frequency-variant.

Abstract: It is possible to perform favorable content reproduction on a reception side. An encoded stream including a plurality of pieces of encoded data having degree-of-priority information is acquired. Among the plurality of pieces of encoded data, decoding processing is performed with respect to a piece of encoded data having a predetermined or more degree of priority, and a decoded stream including decoded data is generated. Further, among the plurality of pieces of encoded data, an encoded stream including a piece of encoded data having less than the predetermined degree of priority is generated. The decoded stream and the encoded stream are simultaneously transmitted as a partially decoded stream to the reception side via a digital interface.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: In general, techniques are described by which to perform spatial relation coding using virtual higher order ambisonic coefficients. A device comprising a memory and a processor may perform the techniques. The memory may be configured to store audio data, the audio data representative of zero-ordered higher order ambisonic (HOA) coefficient, and one or more greater-than-zero-ordered HOA coefficients. The processor may be configured to obtain, based on the one or more greater-than-zero-ordered HOA coefficients, a virtual zero-ordered HOA coefficient. The processor may also be configured to obtain, based on the virtual HOA coefficient, one or more parameters from which to synthesize the one or more greater-than-zero-ordered HOA coefficients. The processor may further be configured to generate a bitstream that includes a first indication representative of the zero-ordered HOA coefficients, and a second indication representative of the one or more parameters.

Abstract: In general, techniques are described by which to perform spatial relation coding of higher order ambisonic coefficients using expanded parameters. A device comprising a memory and a processor may perform the techniques. The memory may be configured to store at least a portion of a bitstream, the bitstream including a first indication representative of an HOA coefficient associated with the spherical basis function having an order of zero, and a second indication representative of one or more parameters. The processor may be configured to perform parameter expansion with respect to the one or more parameters to obtain one or more expanded parameters, and synthesize, based on the one or more expanded parameters and the HOA coefficient associated with the spherical basis function having the order of zero, one or more HOA coefficients associated with one or more spherical basis functions having an order greater than zero.

Abstract: This is provided to achieve capability of avoiding hindrance of accurate reflection of intention at the time of production due to execution of frame interpolation on the reception side. A predetermined container including a video stream obtained by performing encoding operation on moving image data of a predetermined frame rate is transmitted. Information for restricting frame interpolation is inserted into one or both of a layer of the container and a layer of the video stream. For example, the information for restricting frame interpolation includes information for prohibiting frame interpolation. Moreover, for example, the information for restricting frame interpolation includes information indicating the number of times of frame repeats.

Abstract: To enable, on a receiving side, processing obtaining predetermined information to be performed easily and appropriately in a case the predetermined information is divided into a predetermined number of audio frames and transmitted. The predetermined information is inserted into an audio compressed data stream. The audio compressed data stream into which the predetermined information is inserted is transmitted. It is possible to insert each of the pieces of divided information obtained by dividing the predetermined information into the predetermined number of audio frames of the audio compressed data stream. Information indicating the overall size of the predetermined information is added to a first piece of divided information. It is possible to ensure space for storing the predetermined information in a storage medium on the basis of the information indicating the overall size of the predetermined information at a time point where the first piece of divided information is obtained.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. An aspect of the invention further relates to methods and apparatus decoding multiplexed and perceptually encoded HOA signals, including transforming a vector of PCM encoded spatial domain signals of the HOA representation to a corresponding vector of coefficient domain signals by multiplying the vector of PCM encoded spatial domain signals with a transform matrix and de-normalizing the vector of PCM encoded and normalized coefficient domain signals, wherein said de-normalizing comprises.

Abstract: An encoder and an encoding method for a multi-channel signal, and a decoder and a decoding method for a multi-channel signal are disclosed. A multi-channel signal may be efficiently processed by consecutive downmixing or upmixing.

Abstract: An audio content playback method for a portable terminal. The audio content playback method includes checking a channel that is supportable by audio content that is currently engaged in group's simultaneous playback, in group's simultaneous playback of the audio content. The method includes allocating a channel to each of devices included in a group based on position information of each device included in the group or based on an input state in a user interface environment that is preset for channel allocation for each device included in the group, and transmitting the allocated channel information to each device included in the group to allow the device to select its allocated channel and play the audio content.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: A device for processing audio signals includes an interchannel phase difference (IPD) mode selector and an IPD estimator. The IPD mode selector is configured to select an IPD mode based on at least a strength value associated with a temporal misalignment between a first audio signal and a second audio signal. The IPD estimator is configured to determine IPD values based on the first audio signal and the second audio signal. The IPD values have a resolution corresponding to the selected IPD mode.

Abstract: A decoder for generating a frequency enhanced audio signal, includes: a feature extractor for extracting a feature from a core signal; a side information extractor for extracting a selection side information associated with the core signal; a parameter generator for generating a parametric representation for estimating a spectral range of the frequency enhanced audio signal not defined by the core signal, wherein the parameter generator is configured to provide a number of parametric representation alternatives in response to the feature, and wherein the parameter generator is configured to select one of the parametric representation alternatives as the parametric representation in response to the selection side information; and a signal estimator for estimating the frequency enhanced audio signal using the parametric representation selected.

Abstract: The encoding and decoding of HOA signals using Singular Value Decomposition includes forming based on sound source direction values and an Ambisonics order corresponding ket vectors (|Y(?s)) of spherical harmonics and an encoder mode matrix (?OxS). From the audio input signal (|x(?s)) a singular threshold value (?s) determined. On the encoder mode matrix a Singular Value Decomposition is carried out in order to get related singular values which are compared with the threshold value, leading to a final encoder mode matrix rank (rfine). Based on direction values (?l) of loudspeakers and a decoder Ambisonics order (Nl), corresponding ket vectors (|Y(?l)) and a decoder mode matrix (?OxL) are formed. On the decoder mode matrix a Singular Value Decomposition is carried out, providing a final decoder mode matrix rank (rfind).

Abstract: The present subject matter provides a technical solution to the technical problems facing sound localization by separating sounds and reproducing the separated sounds using a set of loudspeakers and a set of headphones. A general soundtrack that is meant to be experienced throughout the room would play through the loudspeakers, and specific sounds that are meant to be experienced near the listener would be played through a binaural representation in the headphones. The headphones may be selected to avoid occluding the ear, allowing sound produced at the loudspeakers to be heard clearly. This separation and reproduction of sounds using a combination of a loudspeaker and headphone provides a technical solution to the technical problem facing typical surround sound systems by localizing sounds for listeners in any location within a room. This improves reproduction accuracy of location-specific audio objects, including audio objects above or below a coplanar speaker configuration.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

Abstract: From an HOA signal representation (c(t)) of a sound field having an order of N and a number 0=(N+1)2 of coefficient sequences a mezzanine HOA signal representation (wMEZZ(t)) is generated that consists of an arbitrary number I<0 of virtual loudspeaker signals WMEZZ1(t), WMEZZ,2(t), . . . , WMEZZ,I(t). 0 directions are computed which are nearly uniformly distributed on the unit sphere. The mode vectors with respect to these directions are linearly weighted for constructing a matrix, of which the pseudo-inverse is used for multiplying the HOA signal representation (c(t)) in order to form (11) the mezzanine HOA signal representation (WME.ZZ(t)).

Abstract: A method for using a loudspeaker array that is housed in a loudspeaker cabinet to present audio content to a listener in a room includes receiving (1) an audio channel that includes audio content and (2) acoustical characteristics of the room. The method also produces (1) a first beamformer input signal from the audio channel and (2) a second beamformer input signal and a third beamformer input signal by decorrelating the audio channel and adjusting the audio channel in accordance with the acoustical characteristics of the room. The second and third beamformer input signals are different de-correlated versions of the audio channel. The method also generates driver signals from the first, second, and third beamformer input signals to drive the loudspeaker array to produce a main beam, a first ambient beam, and a second ambient beam, respectively. Other embodiments are also described and claimed.

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

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: 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: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. A vector of coefficient domain signals is separated into a vector of coefficient domain signals having a constant number of HOA coefficients and a vector of coefficient domain signals having a variable number of HOA coefficients. The constant-number HOA coefficients vector is transformed to a corresponding spatial domain signal vector. In order to facilitate high-quality coding, without creating signal discontinuities the variable-number HOA coefficients vector of coefficient domain signals is adaptively normalized and multiplexed with the vector of spatial domain signals.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Disclosed is an audio signal processing device for processing an audio signal. The audio signal processing device includes a receiving unit configured to receive the audio signal; a processor configured to determine whether to render the audio signal by reflecting a location of a sound image simulated by the audio signal on the basis of metadata for the audio signal, and render the audio signal according to a result of the determination; and an output unit configured to output the rendered audio signal.

Abstract: The present disclosure regards a binaural hearing system configured to receive sound signals from the environment having two hearing instruments to be worn on respective sides of the head of a user and to generate a binaural signal using the received sound signals of both hearing instruments.

Abstract: According to an aspect of an embodiment, an audio generation method includes: receiving an audio signal through at least one microphone; generating an input channel signal corresponding to each of the at least one microphone based on the received audio signal; generating a virtual input channel signal based on the input channel signal; generating additional information including reproduction locations of the input channel signal and the virtual input channel signal; and transmitting a multi-channel audio signal and the additional information, the multi-channel audio signal including the input channel signal and the virtual input channel signal.

Abstract: The encoding and decoding of HOA signals using Singular Value Decomposition includes forming based on sound source direction values and an Ambisonics order corresponding ket vectors (|Y(?s)) of spherical harmonics and an encoder mode matrix (?O×S). From the audio input signal (|x(?s)) a singular threshold value (?S) determined. On the encoder mode matrix a Singular Value Decomposition is carried out in order to get related singular values which are compared with the threshold value, leading to a final encoder mode matrix rank (rfins). Based on direction values (?l) of loudspeakers and a decoder Ambisonics order (Nl), corresponding ket vectors (|Y(?l)) and a decoder mode matrix (?O×L) are formed. On the decoder mode matrix a Singular Value Decomposition is carried out, providing a final decoder mode matrix rank (rfind).

Abstract: Higher Order Ambisonics represents three-dimensional sound independent of a specific loudspeaker set-up. However, transmission of an HOA representation results in a very high bit rate. Therefore compression with a fixed number of channels is used, in which directional and ambient signal components are processed differently. For coding, portions of the original HOA representation are predicted from the directional signal components. This prediction provides side information which is required for a corresponding decoding. By using some additional specific purpose bits, a known side information coding processing is improved in that the required number of bits for coding that side information is reduced on average.

Abstract: A signaling processor is provided. The signaling processor includes a frequency domain processing module configured to generate a cut-off frequency of an input signal and to generate level information for adjusting a level of a high frequency recovery signal and a time domain processing module configured to receive the cut-off frequency and the level information from the frequency domain processing module, to generate a signal having a frequency greater than or equal to the cut-off frequency using part of a signal of a frequency lower than the cut-off frequency in the input signal, to generate the high frequency recovery signal by adjusting a level of the generated signal using the level information, and to synthesize the high frequency recovery signal with the input signal.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: The present document relates to the field of encoding and decoding of audio. In particular, the present document relates to encoding and decoding of an audio scene comprising audio objects. A method (400) for encoding metadata relating to a plurality of audio objects (106a) of an audio scene (102) is described. The metadata comprises a first set (114, 314) of metadata and a second set (104) of metadata. The first and second sets (104, 114, 314) of metadata comprise one or more data elements which are indicative of a property of an audio object (106a) from the plurality of audio objects (106a) and/or of a downmix signal (112) derived from the plurality of audio objects (106a). The method (400) comprises identifying (401) a redundant data element which is common to the first and second sets (104, 114, 314) of metadata.

Abstract: Disclosed are a multi-channel audio signal processing method and a multi-channel audio signal processing apparatus. The multi-channel audio signal processing method may generate N channel output signals from N/2 channel downmix signals based on an N-N/2-N structure.

Abstract: In general, techniques are directed to intermediate compression of higher order ambisonic audio data. For example, a device comprising a processor and a memory may be configured to perform the techniques. The memory may be configured to store an intermediately formatted audio data generated as a result of an intermediate compression of higher order ambisonic audio data. The one or more processors may be configured to process the intermediately formatted audio data.

Abstract: A method for selecting a subset of channels of (e.g., determined from) at least a segment of a multichannel audio program for watermarking and watermarking the selected subset of channels, and a system or device configured to implement any embodiment of the method, or including a buffer which stores at least one frame or other segment of a multichannel audio program generated by any embodiment of the method or steps thereof. Some embodiments generate watermarking metadata during program creation including by analyzing audio content to be included in segments of a multichannel program, determining at least one watermark suitability value for each channel of each of the segments, and including the watermark suitability values (or watermarking data determined therefrom) as metadata in the program. Some embodiments are implemented by a playback system which determines the selected subset of channels to be watermarked.

Abstract: In general, techniques are described for transforming spherical harmonic coefficients. A device comprising one or more processors may perform the techniques. The processors may be configured to parse the bitstream to determine transformation information describing how the sound field was transformed to reduce a number of the plurality of hierarchical elements that provide information relevant in describing the sound field. The processors may further be configured to, when reproducing the sound field based on those of the plurality of hierarchical elements that provide information relevant in describing the sound field, transform the sound field based on the transformation information to reverse the transformation performed to reduce the number of the plurality of hierarchical elements.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Disclosed is a non-transitory computer readable storage medium which receives, by an audio decoder (operating in a specific playback environment different from a reference channel configuration), an audio signal for the reference channel configuration. The audio signal includes audio sample data and encoder-generated loudness metadata which includes a plurality of portions of loudness metadata for a plurality of playback environments. The plurality of portions of loudness metadata includes one or more respective portions of loudness metadata for each playback environment in the plurality of playback environments. The medium also selects one or more portions of specific loudness metadata (based on the specific playback environment), from among the plurality of portions of loudness metadata for the plurality of playback environments.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

Abstract: A device having first means for directing a sound wave to a first side of a target head location; a first means for encircling the first directing means, being configured to phase focus the sound wave to a proximate center point of a left ear of the listener; a second means for directing the sound wave to a second side of the target head location of the listener; a second means for encircling the second directing means, being configured to phase focus the sound wave to a proximate center point of a right ear of the listener; and means for encapsulating the first and second encircling means, said encapsulating means being configured to proximately surround the target head location of the listener.