Abstract: An audio signal processing apparatus configured to process a plurality of audio signals of a shared-volume stereo system is provided. The audio signal processing apparatus includes a plurality of signal processing channels. The signal processing channels are configured to filter the audio signals to obtain high-frequency components and low-frequency components. The signal processing channels perform a signal processing operation on the low-frequency components to generate a low-frequency modulation signal. The signal processing channels sum up the high-frequency components and the low-frequency modulation signal to reproduce a plurality of audio reproduction signals. The signal processing channels drive loudspeakers of the shared-volume stereo system according to the audio reproduction signals. Furthermore, an audio signal processing method is also provided.

Abstract: A multi-rate audio processing system and method provides real-time measurement and processing of amplitude/phase changes in the transition band of the lowest frequency subband caused by the audio processing that can be used to apply amplitude/phase compensation to the higher subband(s). Tone signals may be injected into the transition band to provide strong tonal content for measurement and processing. The real-time measurement and compensation adapts to time-varying amplitude/phase changes regardless of the source of the change (e.g. non-linear time-varying linear or user control parameters) and provides universal applicability for any linear audio processing.

Abstract: A sound reproduction system that is capable of providing at least two separate sound zones in one coherent listening room. In each sound zone, resulting acoustic signals substantially corresponds to a respective audio source signal associated with the same sound zone, and the contribution of audio source signals associated with a different sound zone to the resulting sound signal is minimized.

Abstract: A method of determining acoustical characteristics of a room or venue using a microphone unit having four omni-directional microphones placed at ends of a tetrahedro-mounting unit which are equidistant to a middle point of a mounting unit. The four microphones detect impulse responses for each of n sound sources. The detected impulse responses are analyzed: (1) by determining a direct-sound-component direction, delay, and frequency response; (2) by determining an early-reflection direction, delay, and frequency response of each m early reflection; and (3) in view of late-reverberation components by determining a delay and frequency responses. Direct-sound-transmission-function filter parameters are calculated based on the determined direct-sound-component direction, delay, and frequency response M early-reflection-transmission-function filters parameters are calculated based on the m determined directions, delays, and frequency responses of the m early-reflection components.

Abstract: According to various embodiments, a method for outputting a modified audio signal may be provided. The method may include: receiving from a user an input indicating an angle; determining a parameter for a head-related transfer function based on the received input indicating the angle; modifying an audio signal in accordance with the head-related transfer function based on the determined parameter; and outputting the modified audio signal.

Abstract: In some embodiments, a method for automatic detection of polarity of speakers, e.g., speakers installed in cinema environments. In some embodiments, the method determines relative polarities of a set of speakers (e.g., loudspeakers and/or drivers of a multi-driver loudspeaker) using a set of microphones, including by measuring impulse responses, including an impulse response for each speaker-microphone pair; clustering the speakers into a set of groups, each group including at least two of the speakers which are similar to each other in at least one respect; and for each group, determining and analyzing cross-correlations of pairs of impulse responses (e.g., pairs of processed versions of impulse responses) of speakers in the group to determine relative polarities of the speakers. Other aspects include systems configured (e.g., programmed) to perform any embodiment of the inventive method, and computer readable media (e.g., discs) which store code for implementing any embodiment of the inventive method.

Abstract: An audio system has a first channel for receiving a first input signal and driving a first speaker and a second channel for receiving a second input signal and driving a second speaker. A first feedforward circuit couples an input of the second channel circuit to an input of the first channel circuit. A second feedforward circuit couples an input of the first channel circuit to an input of the second channel circuit. Circuit parameters of the first and the second feedforward circuits are determined such that a first detected output signal is zero when the first input signal is non-zero and the second input signal is zero, and a second detected output signal is zero when the second input signal is non-zero and the first input signal is zero. The audio system is configured to operate using the determined circuit parameters for the first and the second feedforward circuits.

Abstract: Embodiments are described for a system of rendering spatial audio content in a listening environment. The system includes a rendering component configured to generate a plurality of audio channels including information specifying a playback location in a listening area, an upmixer component receiving the plurality of audio channels and generating, for each audio channel, at least one reflected sub-channel configured to cause a majority of driver energy to reflect off of one or more surfaces of the listening area, and at least one direct sub-channel configured to cause a majority of driver energy to propagate directly to the playback location.

Abstract: A method and device are disclosed for determining an inter-channel time difference of a multi-channel audio signal having at least two channels. A determination is made at a number of consecutive time instances, inter-channel correlation based on a cross-correlation function involving at least two different channels of the multi-channel audio signal. Each value of the inter-channel correlation is associated with a corresponding value of the inter-channel time difference. An adaptive inter-channel correlation threshold is adaptively determined based on adaptive smoothing of the inter-channel correlation in time. A current value of the inter-channel correlation is then evaluated in relation to the adaptive inter-channel correlation threshold to determine whether the corresponding current value of the inter-channel time difference is relevant. Based on the result of this evaluation, an updated value of the inter-channel time difference is determined.

Abstract: A digital speaker system that effectively uses the characteristic of a signal processing circuit is constructed. A digital speaker system 1 having plural speakers has a signal processing circuit 10 for outputting plural right-sound digital signals and plural left-sound digital signals, and at least one of the plural speakers is operated to function as a monaural sound speaker 2 to which any one or plural right-sound digital signals output from the signal processing circuit 10 and any left-sound digital signals whose number corresponds to the number of the signals are input to output monaural sounds.

Abstract: In general, techniques are described for grouping audio objects into clusters. In some examples, a device for audio signal processing comprises a cluster analysis module configured to, based on a plurality of audio objects, produce a first grouping of the plurality of audio objects into L clusters, wherein the first grouping is based on spatial information from at least N among the plurality of audio objects and L is less than N. The device also includes an error calculator configured to calculate an error of the first grouping relative to the plurality of audio objects, wherein the error calculator is further configured to, based on the calculated error, produce a plurality L of audio streams according to a second grouping of the plurality of audio objects into L clusters that is different from the first grouping.

Abstract: An audio reproducing apparatus includes: an obtainment unit configured to obtain a stereo audio signal including an L channel signal and an R channel signal; and a control unit configured to (i) generate a first audio signal for a speaker disposed at an upper position in a listening space and a second audio signal for a speaker disposed at a lower position in the listening space, using the L channel signal and the R channel signal and (ii) determine a gain coefficient corresponding to a degree of correlation between the L channel signal and the R channel signal and multiply by the determined gain coefficient at least one of the first audio signal and the second audio signal, to approximate a ratio between energy of sound reproduced from the first audio signal and energy of sound reproduced from the second audio signal to a predetermined value.

Abstract: An audio system including a first speaker and a second speaker that are arranged in front of a predetermined listening position to be substantially bilaterally symmetrical with respect to the listening position; a third speaker and a fourth speaker that are arranged in front of the predetermined listening position to be substantially bilaterally symmetrical with respect to the listening position; a first attenuator that attenuates components that are equal to or less than a predetermined first frequency of an input audio signal; and an output controller that outputs sounds that are based on the input audio signal from the first speaker and the second speaker and outputs sounds that are based on the first audio signal in which components that are equal to or less than the first frequency of the input audio signal from the third speaker and the fourth speaker.

Abstract: Up-sampler generates an up-sampled sound signal from the sound signal. From the up-sampled sound signal, odd-ordered high-harmonic generator generates an odd-ordered high-harmonic, and even-ordered high-harmonic generator generates an even-ordered high-harmonic. Vowel sound detector identifies whether or not the sound signal is vowel sound, and generates a first gain value and a second gain value. First gain controller amplifies or attenuates the odd-ordered high-harmonic based on the first gain value, and outputs the resultant odd-ordered high-harmonic. Second gain controller amplifies or attenuates the even-ordered high-harmonic based on the second gain value, and outputs the resultant even-ordered high-harmonic. Sound signal processing device adds the gain-adjusted odd-ordered high-harmonic and the gain-adjusted even-ordered high-harmonic to the up-sampled sound signal, and outputs the up-sampled sound signal having the high-harmonics added.

Abstract: An audio processing system (100) comprises a front-end component (102, 103), which receives quantized spectral components and performs an inverse quantization, yielding a time-domain representation of an intermediate signal. The audio processing system further comprises a frequency-domain processing stage (104, 105, 106, 107, 108), configured to provide a time-domain representation of a processed audio signal, and a sample rate converter (109), providing a reconstructed audio signal sampled at a target sampling frequency. The respective internal sampling rates of the time-domain representation of the intermediate audio signal and of the time-domain representation of the processed audio signal are equal. In particular embodiments, the processing stage comprises a parametric upmix stage which is operable in at least two different modes and is associated with a delay stage that ensures constant total delay.

Abstract: Methods and systems for enhancing signal quality are disclosed. A method includes receiving buffers of sound samples including a first microphone signal and a second microphone signal from a first and a second microphone; generating a first cardioid shape signal by subtracting a delayed second microphone signal from the first microphone signal; generating a second cardioid shape signal by subtracting the second microphone signal from a delayed first microphone signal; generating a first level output signal based on the first cardioid shape signal; detecting at least one speech and non-speech region of the first level output signal; generating a second level output signal based on the second cardioid shape signal, and at least one of the speech and non-speech regions of the first level output signal; and removing residuals of noise from the first level output signal based on adaptive weights output and generated second level output signal.

Abstract: An audio signal decoder for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information and in dependence on a rendering information has an object parameter determinator. The object parameter determinator is configured to obtain inter-object-correlation values for a plurality of pairs of audio objects. The object parameter determinator is configured to evaluate a bitstream signaling parameter in order to decide whether to evaluate individual inter-object-correlation bitstream parameter values to obtain inter-object-correlation values for a plurality of pairs of related audio objects, or to obtain inter-object-correlation values for a plurality of pairs of related audio objects using a common inter-object-correlation bitstream parameter value.

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 method for processing audio signals for creating a three dimensional sound environment includes: receiving at least one input signal from at least one sound source; creating a simulated signal at least in part on the basis of the received at least one input signal, the simulated signal representing a simulation of at least one input signal reflecting from the ground or floor; and creating an output signal at least partly on the basis of the simulated signal and the at least one received input signal, the output signal including a plurality of audio channels; at least two channels of the audio channels of the output signal representing signals for sound transducers above a listener's ear level at a nominal listening position, and at least two channels of the audio channels of the output signal representing signals for sound transducers below a listeners ear level at a nominal listening position.

Abstract: The present disclosure relates to a connection member for connecting a headset plug with an electronic device, a headset jack of the electronic device and the electronic device. The connection member comprises a pin, wherein the pin communicates with a terminal positioned near an end portion of the headset plug when the connection member is connected with the headset plug. The connection member disclosed in the present disclosure can reduce the length of the headset to be inserted into the headset jack, and reduce the depth of the headset jack, thereby saving space in the electronic device configured with the headset jack.

Abstract: In processing a multi-channel audio signal having at least three original channels, a first downmix channel and a second downmix channel are provided, which are derived from the original channels. For a selected original channel, channel side information are calculated such that a downmix channel or a combined downmix channel including the first and the second downmix channels, when weighted using the channel side information, results in an approximation of the selected original channel. The channel side information and the first and second downmix channels form output data to be transmitted to a decoder, which, in case of a low level decoder only decodes the first and second downmix channels or, in case of a high level decoder provides a full multi-channel audio signal based on the downmix channels and the channel side information.

Abstract: A method includes, for each of a number of subbands of a frequency range and for at least first and second frequency-domain signals that are frequency-domain representations of corresponding first and second audio signals: determining a time delay of the first frequency-domain signal that removes a time difference between the first and second frequency-domain signals in the subband. The method includes forming a first resultant signal including, for each of the number of subbands, a sum of one of the first or second frequency-domain signals shifted by the time delay and of the other of the first or second frequency-domain signals; and forming a second resultant signal including, for each of the number of subbands, a difference between the shifted one of the first or second frequency-domain signals and the other of the first or second frequency-domain signals. Apparatus and program products are also disclosed.

Abstract: An apparatus comprising: an audio signal analyser configured to analyse at least one audio signal to determine at least one audio component with an associated orientation parameter; a reference definer configured to define at least one of: a reference orientation for an apparatus; and a reference position for the apparatus; a directional determiner configured to determine a direction value based on the reference orientation/position for the apparatus and at least one of: an orientation of the apparatus; a position of the apparatus; an orientation of a further apparatus co-operating with the apparatus; and a position of the further apparatus; and a directional processor configured to process at least one associated directional parameter for the at least one audio component dependent on the direction value.

Abstract: An equalizer that linearly interpolates between two equalization states when transitioning from one equalization state to the other equalization state is described. The equalizer includes a transfer function generator and an equalization module. Each equalization state is defined or determined based on a set of parameters. The transfer function generator generates a set of interpolated transfer functions by performing linear interpolation on a first equalization state and a second equalization state based on the set of parameters. The linear interpolation is performed on corresponding Z-domain poles and zeros of the transfer functions of the first and second equalization states. The equalization module applies the set of interpolated transfer functions generated by the transfer function generator to an input audio signal.

Abstract: This document discusses, among other things, systems and methods to communicate data over a data bus during a first period of a clock signal with a uniform power distribution, including providing a complimentary bit state of the data during a first portion of the first period of the clock signal and providing an actual bit state of the data during a second portion of the first period of the clock signal. In an example, the first period can include first, second, third, and fourth portions, and the systems and methods can include providing a complimentary bit state of the data during first and fourth portions of the first period of the clock signal and an actual bit state of the data during a second portion of the first period of the clock signal.

Abstract: A directivity control apparatus controls a directivity of a sound collected by a first sound collecting unit including a plurality of microphones. The directivity control apparatus includes a directivity forming unit, configured to form a directivity of the sound in a direction toward a monitoring target corresponding to a first designated position in an image displayed on a display unit, and an information obtaining unit, configured to obtain information on a second designated position in the image displayed on the display unit, designated in accordance with a movement of the monitoring target. The directivity forming unit is configured to change the directivity of the sound toward the monitoring target corresponding to the second designated position by referring to the information on the second designated position obtained by the information obtaining unit.

Abstract: A method and system for creating and editing video and/or audio tracks is described. The method includes providing at least one artist, venue, and track available for selection and providing at least one clip associated with the at least one artist, venue, and track. The method also includes allowing a user to create a custom track from the at least one clip. The system includes a plurality of video cameras for recording a live performance at a plurality of positions. The system also includes at least one server for storing a plurality of video clips created from the plurality of video cameras and an application stored on the at least one server for allowing a user to access the plurality of video clips via the Internet.

Abstract: A multiband limiter with selective sideband linking includes first and second frequency band splitters, a first and second plurality of limiters, first and second summers, and a plurality of selectable links coupling the first plurality of limiters to the second plurality of limiters. The first plurality of limiters each have a band input coupled to one of the first plurality of band outputs, a link port and a limiter output, and the first summer is receptive to the limiter outputs of the first plurality of limiters and has a first channel output. The second plurality of limiters each have a band input coupled to one of the second plurality of band outputs, a link port and a limiter output, and the second summer is receptive to the limiter outputs of the second plurality of limiters and has a second channel output.

Abstract: Systems and methods for enhancing the low frequency response of a loudspeaker for relatively low input level audio signals and protect the loudspeaker for relatively high input level audio signals make use of a crossover network configured so separate an audio input signal into at least two frequency bands including a low frequency band; and a signal compressor responsive to the energy level of the low frequency portion of an input audio signal in the low frequency band and configured to provide amplification gain on the low frequency portion of the input signal when the energy level of the low frequency portion of the input signal is relatively low. One or more peak limiters may be positioned before the summer(s), in the circuit, but after the low band compressor. The EQ unit(s) may be positioned before all processing to the L and R inputs.

Abstract: Systems and methods facilitating removal of content from audio files are described. A method includes identifying a sound recording in a first audio file, identifying a reference file having at least a defined level of similarity to the sound recording, and processing the first audio file to remove the sound recording and generate a second audio file. In some embodiments, winner-take-all coding and Hough transforms are employed for determining alignment and rate adjustment of the reference file in the first audio file. After alignment, the reference file is filtered in the frequency domain to increase similarity between the reference file and the sound recording. The frequency domain representation (FR) of the filtered version is subtracted from the FR first audio and the result converted to a time representation of the second audio file. In some embodiments, spectral subtraction is also performed to generate a further improved second audio file.

Abstract: A test system for testing directional capabilities of an audio device includes a horizontal linear actuator and a vertical linear actuator. The horizontal linear actuator supports a rotary actuator upon which the audio device is placed. The vertical linear actuator supports a sound source. To test the device, the actuators are controlled to establish multiple relative positions of the audio device and the sound source. A test sound is emitted at each of the relative positions. The audio device is configured to provide data generated in response to the test sound at each of the relative positions. The test system receives and records the data along with coordinates indicating the relative positions to which the data corresponds.

Abstract: The present invention discloses a method and device for controlling speaker array sound field based on a quadratic residue sequence combination. The method comprises steps of: (1) fragmenting a designated quadratic residue sequence in terms of the number of array elements, to generate a plurality of quadratic residue subsequences; (2) designing an optimal array phase delay vector utilizing these subsequences; (3) controlling transmission signals of multi-element channels according to the optimal phase delay vector to adjust phase delay; (4) sending the multi-channel signals subjected to adjustment to a multi-channel power amplifier, to drive the speaker array to generate uniform sound field. The device comprises a sound source, an optimal phase delay estimator, an optimal phase delay controller, a multi-channel power amplifier and a speaker array. The invention can expand the coverage range of sound field radiated from an array and improve uniformity of the sound field.

Abstract: A method comprising the steps of: determining one or more impedance values of a loudspeaker; and determining the polarity of a loudspeaker based on the one or more impedance values.

Abstract: A system and method for acoustically reproducing at least two electrical audio signals and establishing at least two sound zones that are represented by individual patterns of reception sound signals includes processing the at least two electrical audio signals to provide processed electrical audio signals; converting the processed electrical audio signals into corresponding acoustic audio signals with at least two loudspeakers that are arranged at positions separate from each other; transferring each of the acoustic audio signals according to a transfer matrix from each of the loudspeakers to each of the sound zones where they contribute to the reception sound signals; and processing of the at least two electrical audio signals comprises inverse filtering according to a filter matrix. Inverse filtering is configured to compensate for the room transfer matrix so that each one of the reception sound signals corresponds to one of the electrical audio signals.

Abstract: An apparatus for generating an enhanced downmix signal on the basis of a multi-channel microphone signal has a spatial analyzer configured to compute a set of spatial cue parameters having a direction information describing a direction-of-arrival of a direct sound, a direct sound power information and a diffuse sound power information on the basis of the multi-channel microphone signal. The apparatus also has a filter calculator for calculating enhancement filter parameters in dependence on the direction information describing the direction-of-arrival of the direct sound, in dependence on the direct sound power information and in dependence on the diffuse sound power information. The apparatus also has a filter for filtering the microphone signal, or a signal derived therefrom, using the enhancement filter parameters, to obtain the enhanced downmix signal.

Abstract: Systems and methods in accordance with various embodiments of the present disclosure overcome one or more deficiencies in conventional approaches to stereo playback. In particular, various embodiments attempt to cancel or reduce the sound distortion and/or noise from “crosstalk signals” such that stereo effect can be maintained and/or enhanced. In some embodiments, the various embodiments attempt to reduce and/or compensate for the loss of low frequency (bass) sound signals. Moreover, a listener's position, such as his/her head position, can be tracked such that the enhanced stereo playback can be maintained if the listener changes position.

Abstract: A method for conversion of a n-channel audio signal (L, R, Ls, Rs) into a two-channel audio signal (Ro, Lo), where n?4 and integer, includes the step of generating either one of the two-channel audio signals, right (Ro) or left (Lo), by a combination of: a front (R, L) and rear (Rs, Ls) signal components of the n-channel audio signal of the same side (right or left), and a front (L, R) signal component of the n-channel audio signal of the other side (left or right), and a term dependent of n.

Abstract: This application relates to a system for compression and mixing for hearing assistance devices by application of compression to individual sound sources before mixing, according to one example. Variations of the present system using surround sound provide separate signals from a surround sound synthesizer which are compressed prior to mixing of the signals.

Abstract: The present invention provides an arrangement, e.g. a method, an apparatus and a computer program. The arrangement comprises obtaining information indicative of spatial positions of one or more sound sources within an audio image of a composite audio signal, obtaining information indicative of the types of the one or more sound sources, determining, for each of the one or more sound sources, a position in a display of the apparatus indicative of the spatial position of the sound source and causing an apparatus to display an item representing the sound source in the determined position in the display, wherein the item representing the sound source is determined on basis of the type of the sound source.

Abstract: Techniques are presented for creating multichannel output signals from input audio signals. A first signal is determined based on a number of subbands into which the input audio signals are divided and based at least in part on a directional estimation wherein the subbands having dominant sound source directions are emphasized relative to subbands having directional estimates that deviate from directional estimates of the dominant sound source directions. A second signal is determined based on the number of subbands wherein an ambient component is introduced to create a perception of an externalization for a sound image. A resultant audio signal is created using the first and second signals. The resultant audio signal is one of a number of multichannel signals. Additionally, it is determined whether binaural audio output or multichannel audio output (or both) is to be output, and the appropriate number of audio output signals are determined and output.

Abstract: A method of generating a stereoscopic image registration and color balance evaluation display enables a user to align two cameras used to produce a stereoscopic video signal or to analyze a received stereoscopic video signal. Left and right images from the cameras are converted into desired video signal components. Corresponding pixels from the left and right images are compared for each of the video signal components, and plotted as x and y inputs to a Cartesian plot to produce respective two-dimensional (2D) histograms. The multiple 2D histograms are presented as a Quad Diamond display so a user may align the two cameras or analyze the received stereoscopic video signal.

Abstract: A control circuit coupled to a differential audio output stage may, responsive to a transition in a power supply voltage generated by a power supply, modify at least one of: (i) a first bandwidth associated with the power supply; (ii) a second bandwidth associated with a common-mode voltage generator for generating a desired output common-mode voltage based on the power supply voltage; and (iii) a third bandwidth associated with a common-mode feedback loop of the audio-output stage for setting an actual common-mode voltage at each of the pair of differential output terminals based on the desired output common-mode voltage; such that the second bandwidth is greater than or substantially equal to the first bandwidth during the transition and the third bandwidth is greater than or substantially equal to the second bandwidth during the transition.

Abstract: A sound generator includes a diaphragm including n sub-diaphragms (n, the amount of the sub-diaphragms); n voice coils corresponding to the sub-diaphragms; a signal source for outputting signals of n channels; n high-pass filters for receiving the signals of n channels and outputting n high frequency signals; n low-pass filters for receiving the signals of n channels and outputting n low frequency signals; a first mixer for mixing the n low frequency signals and then outputting a low frequency signal; and n second mixers for mixing the low frequency signal and the n high frequency signals, and then outputting n driving signals. The n driving signals actuates the n corresponding sub-diaphragms for producing stereo sounds.

Abstract: A processing method and processing apparatus suitable for stereo audio output enhancement. The processing apparatus can include an input portion configurable to receive a set of input signals, an intermediate portion coupled to the input portion and an output portion coupled to the intermediate portion. The input portion can be configured to produce processed input signals based on the set of input signals. The intermediate portion can be configured to produce a compensated signal based on the processed input signals. The intermediate portion can also be configured to produce a first mixed signal and a second mixed signal based on the set of input signals and at least a portion of the compensated signal. The output portion can be configured to produce a set of output signals based on the first and second mixed signals.

Abstract: An audio processing method and an audio processing apparatus are described. A mono-channel audio signal is transformed into a plurality of first subband signals. Proportions of a desired component and a noise component are estimated in each of the subband signals. Second subband signals corresponding respectively to a plurality of channels are generated from each of the first subband signals. Each of the second subband signals comprises a first component and a second component obtained by assigning a spatial hearing property and a perceptual hearing property different from the spatial hearing property to the desired component and the noise component in the corresponding first subband signal respectively, based on a multi-dimensional auditory presentation method. The second subband signals are transformed into signals for rendering with the multi-dimensional auditory presentation method.

Abstract: Methods are disclosed for improving sound localization of the human ear. In some embodiments, the method may include creating virtual movement of a plurality of localized sources by applying a periodic function to one or more location parameters of a head related transfer function (HRTF).

Abstract: The present technology relates to an acoustic signal processing apparatus, an acoustic signal processing method, a program, and a recording medium for improving a localization of sound of a sound image at a position deviated from the front center plane of a listener to the left side of the right side. A binauralization processing unit generates a first binaural signal by superimposing an HRTF on an opposite side of a sound source on an acoustic signal and a second binaural signal by attenuating components of a band in which a first notch and a second notch of the HRTF on the opposite side of the sound source appear among components of a signal obtained by superimposing an HRTF on a side of the sound source on the acoustic signal. A crosstalk compensation processing unit performs, with respect to the first binaural signal and the second binaural signal, a crosstalk compensation for canceling out an acoustic transfer characteristic and a crosstalk.

Abstract: A speaker apparatus includes speaker units, a first inputting section to which one channel audio signal is supplied, a second inputting section to which at least L-channel and R-channel audio signals are supplied, an acoustic effect imparting section that performs a first signal processing on the audio signal supplied to the first inputting section to output L channel and R channel audio signals, an outputting section that outputs the audio signal supplied to the first inputting section to at least one of the speaker units, and a widening effect imparting section that performs a second signal processing for imparting a sound image widening effect on both the L-channel and R-channel audio signals and the at least L-channel and R-channel audio signals and supplies processed audio signals to speaker units for corresponding channels respectively.

Abstract: The invention is related to a data structure for Higher Order Ambisonics HOA audio data, which data structure includes 2D or 3D spatial audio content data for one or more different HOA audio data stream descriptions. The HOA audio data can have on order of greater than ‘3’, and the data structure in addition can include single audio signal source data and/or microphone array audio data from fixed or time-varying spatial positions.

Abstract: A method for processing audio signals can include receiving left and right front audio signals and left and right rear audio signals, where the left and right rear audio signals. In addition, the method can include applying at least one front perspective filter to each of the left and right front audio signals to yield filtered left and right front output signals, where the left and right front output signals each drive a front speaker. Moreover, the method can include applying at least one rear perspective filter to each of the left and right rear audio signals to yield left and right rear output signals, where the left and right rear output signals each drive a rear speaker to simulate a rear surround sound effect when positioned in front of a listener.