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: 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: 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: 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: 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 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: 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.

Abstract: The invention concerns sound spatialization with multichannel encoding for binaural reproduction on two loudspeakers, the spatial encoding being defined by encoding functions associated with multiple encoding channels and the decoding by applying filters for binaural reproduction.

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: An analog input stage has m differential input channels, wherein m>1. The analog input stage is configured to select one of the m differential input channels and provide an output signal. The analog input stage has n identical selection units each having m differential channel inputs and one differential output, wherein n is at least 2m?1. Each selection unit is operable to be coupled to any of the differential input channels through respective differential multiplexer units, wherein the multiplexor units are driven to select one of the differential input channels and couple the selected differential channel input through a butterfly switch unit with the differential output of the selection unit. The differential output signals of the n selection units are combined whereby unwanted crosstalk from channels other than a selected channel are removed by cancellation.

Abstract: A method of enhancing vertical polar localization of a head related transfer (HRTF). The method includes splitting an audio signal and generating left and right output signals by determining a log lateral component of the respective frequency-dependent audio gain that is equal to a median log frequency-dependent audio gain for all audio signals of that channel having a desired perceived source location. A vertical magnitude of the respective audio signal is enhanced by determining a log vertical component of the respective frequency-dependent audio gain that is equal to a product of a first enhancement factor and a different between the respective frequency-dependent audio gain at the desired perceived source location and the lateral magnitude of respective audio signal. The output signals are time delayed according to an interaural time.

Abstract: A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

Abstract: A system for simultaneous visual data presentation is provided having a processor and memory with instructions for execution by the processor for: receiving attended data; receiving unattended data; and generating presentation data. The presentation is suitable for simultaneous presentation of the attended data and the unattended data in a manner in which the unattended data at least partially overlaps the attended data and the unattended data does not interfere with the attended data by replacing, erasing, or suppressing the attended data, and vice versa. The presented unattended data gives cues that the user can sense and provide information to the user while attending to the presented attended data, without attending to the presented unattended data.

Abstract: An encoding device, a decoding device, and encoding and decoding methods are provided, wherein when a multi-channel signal is encoded with high efficiency, using an adaptive filter, the number of arithmetic operations to update a filter coefficient of the adaptive filter can be reduced. An update range determination unit determines the range of a filter coefficient order (update order range) of a filter coefficient to be updated, among filter coefficients gk(n) of the adaptive filter, on the basis of a mutual correlation function between an input (L) signal and an input (R) signal. The adaptive filter updates the filter coefficient gk(n) of the filter coefficient order (n) to be updated, using a decoding (L) signal and a decoding error (R) signal.

Abstract: An audio or video enhancement module incorporates advanced audio enhancement effects. These advanced effects cover a wide range including but not limited to 3D audio spatialization (virtualization) and transient enhancement. The device is configured for receiving an audio input signal, in analog or digital form, enhancing the received signal, and transmitting an output audio signal having the enhanced effects embedded in the signal. The output signal may be either digital or audio. Further, the output signal may be a mono signal, a stereo signal, a MIDI signal, or a multichannel signal (such as 2.1, 5.1, 7.1, etc.). User input to the module is provided to control the amount of audio enhancement.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for stereophonic acoustic echo cancellation. The method includes collecting, at a same time, a first audio sample of an audio source from a first omnidirectional microphone and a second audio sample of the audio source from a second omnidirectional microphone. The method includes delaying the second audio sample by a first amount of time to yield a delayed second audio sample and combining the delayed second audio sample with the first audio sample to produce a first channel, then delaying the first audio sample by a second amount of time to yield a delayed first audio sample and combining the delayed first audio sample with the second audio sample to produce a second channel. Then the method includes outputting the first channel and the second channel as a stereo audio signal of the audio source.

Abstract: A depth processing system can employ stereo speakers to achieve immersive effects. The depth processing system can advantageously manipulate phase and/or amplitude information to render audio along a listener's median plane, thereby rendering audio along varying depths. In one embodiment, the depth processing system analyzes left and right stereo input signals to infer depth, which may change over time. The depth processing system can then vary the phase and/or amplitude decorrelation between the audio signals over time to enhance the sense of depth already present in the audio signals, thereby creating an immersive depth effect.

Abstract: An audio system for a vehicle includes a directional loudspeaker that is mounted to a vehicle seat. The directional loudspeaker radiates a first channel audio signal so that the direction toward an intended location of a first ear position of an occupant of the vehicle seat is a high radiation direction and radiates a second channel audio signal so that the direction toward an intended location of a second ear position of the occupant of the vehicle seat is a high radiation direction. A forward mounted loudspeaker radiates at least one of the first channel audio signal and the second channel audio signal. Signal processing circuitry modifies (e.g., delays) the first channel audio signal to at least one of the directional loudspeaker and the forward mounted loudspeaker to cause one of those speakers to dominate spatial perception.

Abstract: Embodiments are described for a method and system of rendering and playing back spatial audio content using a channel-based format. Spatial audio content that is played back through legacy channel-based equipment is transformed into the appropriate channel-based format resulting in the loss of certain positional information within the audio objects and positional metadata comprising the spatial audio content. To retain this information for use in spatial audio equipment even after the audio content is rendered as channel-based audio, certain metadata generated by the spatial audio processor is incorporated into the channel-based data. The channel-based audio can then be sent to a channel-based audio decoder or a spatial audio decoder. The spatial audio decoder processes the metadata to recover at least some positional information that was lost during the down-mix operation by upmixing the channel-based audio content back to the spatial audio content for optimal playback in a spatial audio environment.

Abstract: An apparatus for enhancing a multichannel audio signal comprising at least two channels configured to: estimate a value representing a direction of arrival associated with a first audio signal from at least a first channel and a second audio signal from at least a second channel of at least two channels of a multichannel audio signal; determine a scaling factor dependent on the direction of arrival associated with the first audio signal and the second audio signal; and apply the scaling factor to a parameter associated with a difference in audio signal levels between the first audio signal and the second audio sign.

Abstract: Methods and systems for processing an audio signal are provided. The method includes generating a pseudorandom sequence and generating at least one reciprocal of the pseudorandom sequence such that the at least one reciprocal is substantially decorrelated with the pseudorandom sequence. The pseudorandom sequence and the at least one reciprocal form a set of sequences. The method further includes convolving the audio signal with the set of sequences to generate a corresponding number of output signals and providing the number of output signals to a corresponding number of loudspeakers.

Abstract: A crosstalk cancellation system and method is disclosed for use in a multi-channel communication system. Crosstalk which couples between channels is cancelled through use of an in-line FFE filter and in-line delay. A cross-connect system associated with each channel includes a cross-connect delay and cross-connect filter. The cross-connect system generates a cancellation signal for each of the channels, which is routed into a junction. The junction subtracts the cancellation signal from the received signal, which has also been delayed and filtered, to remove unwanted cross-talk. During training, cancellation magnitude is monitored at various delay offsets to determine which offset and corresponding filter coefficients, for each delay, maximizes cancellation. The filters are set with filter coefficients that maximize cancellation. The cross-connect delay with the maximum offset is set to zero and its calculated offset amount is established as the in-line delay offset.