134results about "Pseudo-stereo systems" patented technology

This application provides a display device and its audio output method. The display device includes a display screen, a first speaker, and a second speaker. The first speaker is located behind the display screen, and its sound emission direction is towards the upper rear of the display device. The second speaker's sound emission direction is towards the front of the display device or towards the lower part of the display device. The first speaker and the second speaker emit sound asynchronously. The display device of this application has good stereo sound effects.

A delay estimation method includes determining a cross-correlation coefficient of a multi-channel signal of a current frame, determining a delay track estimation value of the current frame based on buffered inter-channel time difference information of at least one past frame, determining an adaptive window function of the current frame, performing weighting on the cross-correlation coefficient based on the delay track estimation value of the current frame and the adaptive window function of the current frame, to obtain a weighted cross-correlation coefficient, and determining an inter-channel time difference of the current frame based on the weighted cross-correlation coefficient.

The present disclosure relates to "spatial audio controllers." The present disclosure provides a method performed by a local device communicatively coupled with a number of remote devices, the method comprising: receiving, from each remote device in a communication session with the local device, an input audio stream; receiving, for each remote device, a set of parameters; determining, for each input audio stream, based on the set of parameters, whether the input audio stream is to be 1) rendered individually, or 2) rendered as a mix of input audio streams; for each input audio stream determined to be rendered individually, spatially rendering the input audio stream as an individual virtual sound source containing only the input audio stream; and for input audio streams determined to be rendered as a mix of input audio streams, spatially rendering the mix of input audio streams as a single virtual sound source containing the mix of input audio streams.

The present invention addresses the problem of providing an "audio signal processing device" that achieves a good sense of localization with respect to a sound having a narrow frequency band. [Solution] The present invention is provided with signal processing units provided so as to correspond to respective channels of audio signals of a plurality of channels, and each signal processing unit has: a target sound separation unit (411) that separates a target sound signal indicating a target sound, which is a sound to be subjected to localization enhancement, from a source sound signal, which is an audio signal of the corresponding channel; a sense-of-localization-enhanced sound generation unit that generates a sense-of-localization-enhanced sound signal in which the frequency of the target sound signal separated by the target sound separation unit (411) is multiplied by k (k is an integer of 2 or more); and a synthesis unit that synthesizes and outputs the localization-enhanced sound signal generated by the localization-enhanced sound generation unit and the source sound signal.

Audio signal processor for generating a two-channel audio signal, comprising: an input interface for providing single-channel acoustic data describing an acoustic environment; a two-channel synthesizer for synthesizing two-channel acoustic data from the single-channel acoustic data using a listener position or rotation; and a sound generator for generating the two-channel audio signal from an audio signal and the two-channel acoustic data, wherein the two-channel synthesizer is configured to separate the single-channel acoustic data into at least two parts consisting of a direct sound part and at least one of an early reflection part and a late reverberation part, and to individually process the at least two parts for generating two-channel acoustic data for each part, to determine a separation time instant in the single-channel acoustic data between the direct sound part and the early reflection part or between the early reflection part and the late reverberation part, to extend at least one of the two parts of the separation time instant by a certain number of samples, so that an overlap at the separation time instant is achieved, and to window the at least one extended part using a certain window function accounting for the sample extension.

A method of using an earphone set to deliver binaural beats to a user (e.g., for altering a state of consciousness) and an earphone set 100 therefore are provided. The method involves simultaneously p

The embodiment of the application provides an audio processing method and device, the method comprises the following steps: obtaining M first audio signals obtained by processing a to-be-processed audio signal through M first virtual loudspeakers and N second audio signals obtained by processing the to-be-processed audio signal through N second virtual loudspeakers; obtaining M first HRTFs and N second HRTFs, the M first HRTFs are all HRTFs with a left ear position as a center, and the N second HRTFs are all HRTFs with a right ear position as a center; the M first HRTFs correspond to the M first virtual loudspeakers in a one-to-one manner, and the N second HRTFs correspond to the N second virtual loudspeakers in a one-to-one manner; obtaining a first target audio signal according to the M first audio signals and the M first HRTFs; and obtaining a second target audio signal according to the N second audio signals and the N second HRTFs. The method provided by the embodiment of the application improves the quality of the audio signal output by the audio signal receiving end.

A computer-implemented method for determining audio signal processing parameters for a multi-channel audio system including a plurality of speakers is disclosed. The method involves obtaining at least one first audio response at a first listening position and at least one second audio response at a second listening position, where each audio response corresponds to a respective channel audio signal output by a respective speaker over a predetermined frequency range. The audio signal processing parameters are determined based on a similarity metric calculated between the first and second audio responses over at least a part of the predetermined frequency range. The determined audio signal processing parameters are then provided for further processing of at least one of the channel audio signals, enabling optimization of the audio system's performance across multiple listening positions.

To provide an "audio signal processor" for down-mixing a surround audio signal so as to obtain an excellent sound image feeling.SOLUTION: This device is provided with a pseudo surround audio signal generating part 1 for generating a surround audio signal SigA of 7.1. 2ch from an inputted audio stereo signal SigIN of 4ch, a signal processing part 2 for processing the surround audio signal SigA and outputting it as a signal SigB for down mix, and a down mix part 3 for synthesizing the signal SigB for down mix and outputting a stereo audio signal SigOUT of 2ch. The signal processing unit 2 combines the input audio signal of each channel of the SigA with the audio signal obtained by convolving the head-related transfer function from the speaker corresponding to the audio signal to the listener at a ratio corresponding to the position of the speaker, and outputs the combined signal as the audio signal of the SigB.SELECTED DRAWING: Figure 1

A sound providing method using an earphone set (100, 200, 300, 400) and an earphone set are provided. The sound providing method using the earphone set includes simultaneously providing a first sound wave (W12) to a left ear (EL) of a user by using a left ear earphone (110) of an earphone set and a second sound wave (W14) to a right ear (ER) of the user by using a right ear earphone (120) of the earphone set There is a frequency difference between a first frequency of the first sound wave (W12) and a second frequency of the second sound wave (W14). The frequency difference automatically and cyclically changes along with time.

Some disclosed examples relate to converting input audio data in an input audio format to output audio data in an output audio format. The output audio format may have a higher resolution than the input audio format. The conversion may involve applying a bias decode matrix to the input audio data. The bias decoding matrix may be biased according to an estimated energy distribution of the input audio data and may include a combination of a constant matrix and a variable matrix. The variable matrix may be a covariance matrix corresponding to the input audio data. The bias decode matrix may vary over time as a function of a covariance matrix.

This application provides a method and an apparatus for determining a virtual speaker set. The method for determining a virtual speaker set includes: determining a target virtual speaker from F preset virtual speakers based on a to-be-processed audio signal, where each of the F virtual speakers corresponds to S virtual speakers, F is a positive integer, and S is a positive integer greater than 1; and obtaining, from a preset virtual speaker distribution table, respective position information of S virtual speakers corresponding to the target virtual speaker, where the virtual speaker distribution table includes position information of K virtual speakers, the position information includes an elevation angle index and an azimuth angle index, K is a positive integer greater than 1, F≤K, and F×S≥K. This application can improve audio signal playback effect.

This disclosure relates to a "spatial audio controller". This disclosure provides a method performed by a local device communicatively coupled to a plurality of remote devices, the method comprising: receiving input audio streams from each remote device in a communication session with the local device; receiving a set of parameters for each remote device; for each input audio stream, determining, based on the set of parameters, whether the input audio stream is to be 1) rendered individually or 2) rendered as a mixture of input audio streams; for each input audio stream determined to be rendered individually, spatially rendering the input audio stream as an individual virtual sound source containing only the input audio stream; and for an input audio stream determined to be rendered as a mixture of the input audio streams, spatially rendering the mixture of the input audio streams as a single virtual sound source containing the mixture of the input audio streams.

An audio signal processor for generating a two-channel audio signal, comprising: an input interface (100) for providing single-channel acoustic data describing an acoustic environment; a two-channel combiner (200) for synthesizing two-channel acoustic data from single-channel acoustic data using listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from an audio signal and two-channel acoustic data, wherein the two-channel combiner (200) is configured to separate the single-channel acoustic data into at least two parts consisting of a direct sound portion and at least one of an early reflection portion and a late reverberation portion (210), and to process at least two parts individually for each part to generate two-channel acoustic data (220, 230, 240), and the two-channel combiner (200) is configured to compute two-channel acoustic data for the early reflection portion using a specular reflection portion describing a separate early reflection and a diffusion portion describing the diffusion effect within the early reflection portion (230).

To provide a system that includes one or more computing devices that encode spatial perceptual cues into a monaural channel to generate a plurality of output channels.SOLUTION: A computing device determines a target amplitude response for the mid and side channels of the plurality of output channels, defining a spatial perception associated with one or more frequency-dependent phase shifts. The computing device determines a transfer function of a single-input, multi-output all-pass filter based on the target amplitude response, determines coefficients of the all-pass filter based on the transfer function, and processes the monaural channel with the coefficients of the all-pass filter to generate the plurality of channels having the encoded spatial perceptual cues. The all-pass filter is configured to be colorless with respect to the individual output channels, allowing the placement of spatial cues into the audio stream to be decoupled from the overall coloration of the audio.SELECTED DRAWING: Figure 6

An emulation system (e.g., 100) and method for emulating a listener's audio experience of a candidate loudspeaker speaker system in an acoustic (e.g., residential) space (e.g., 102) includes acquiring binaural impulse responses associated with that listening experience generating associated binaural impulse response ("BIR") filters, and applying them in real time to any user-selected program material for playback over specially prepared headphones (e.g., 900). Headphone listening "modes" may be offered to listeners which emulate the speaker-room experience, and headphones so prepared may be offered to prospective customers of the candidate loudspeaker system (112) whose in-room performance has been emulated. In another embodiment, binaural recordings of automotive audio systems may be used for emulating such systems over headphones.

A processing load at a receiving side is reduced in a case where a plurality of classes of audio data is transmitted. A predetermined number of audio streams including coded data of a plurality of groups is generated and a container of a predetermined format having this predetermined number of audio streams is transmitted. Command information for creating a command specifying a group to be decoded from among the plurality of groups is inserted into the container and / or the audio stream. For example, a command insertion area for the receiving side to insert a command for specifying a group to be decoded is provided in at least one audio stream among the predetermined number of audio streams.

A method for rendering an audio element (e.g., a spatially-heterogeneous audio element), wherein the audio element has an extent and is represented using a set of virtual loudspeakers comprising a middle virtual loudspeaker. The method includes, based on a position of a listener, selecting a position for the middle virtual loudspeaker and / or calculating an attenuation factor for the middle virtual loudspeaker.

An audio headset receives one or more audio signals carrying one or more audio channels and processes the audio channels to generate stereo signals for output to a left and a right speaker of the audio headset. The processing determines a number of the audio channels carried in the received audio signal(s), adjusts level(s) of the audio channels based on the determined number of audio channels and / or adjusts gain and / or phase of the audio channels to control a perceived location of a listener wearing the headset relative to a source of sounds carried in the stereo signals.

Improved tools for authoring and rendering audio reproduction data are provided. Some such authoring tools allow audio reproduction data to be generalized for a wide variety of reproduction environments. Audio reproduction data may be authored by creating metadata for audio objects. The metadata may be created with reference to speaker zones. During the rendering process, the audio reproduction data may be reproduced according to the reproduction speaker layout of a particular reproduction environment.

An apparatus is disclosed for managing and / or reducing harmonic distortion components arising along a signal path associated with an audio signal or audio system. The apparatus comprises a phase generator for generating at least one phase-difference signal or a reference audio signal generated by the phase generator, wherein the or each constant phase difference is adapted to provide cancellation of the harmonic distortion components arising along the signal path, respective amplifier channels for receiving and separately amplifying the audio signal acting as a reference audio signal, respective loudspeaker channels for receiving and separately producing sound corresponding to the amplified audio wherein each loudspeaker channel has substantially equal performance parameters and is adapted to radiate the sound relative to other loudspeaker channels to produce a combined sound that corresponds to the audio signal with harmonic distortion components that are reduced compared to the harmonic distortion components arising along the signal path.

This application provides a scene audio signal encoding method and apparatus. The scene audio signal encoding method in this application includes: obtaining a to-be-encoded scene audio signal, where the scene audio signal includes audio signals of C channels, and C is a positive integer; performing transient detection on M channels, among the C channels, that need transient detection, to obtain transient identifiers of the M channels, where the transient identifier indicates whether a corresponding channel includes a transient signal, and 1≤M≤C; and encoding the transient identifiers of the M channels and the scene audio signal to obtain a bitstream. In this application, a transient signal in the scene audio signal can be processed, to improve quality of a reconstructed audio signal and auditory experience of a user.

The present technology relates to audio processing. An acquisition unit acquires metadata including position information indicative of a position of an audio object and sound image information configured from a vector of three dimensions and representative of an extent of a sound image from the position. A vector calculation unit determines a three-dimensional vector representative of the position of the audio object, wherein a start point of the three-dimensional vector is the origin in a three-dimensional coordinate system whose origin is given by a position of a user. The vector calculation unit further calculates, based on the three-dimensional vector and a ratio between a horizontal direction angle and a vertical direction angle of a region representative of the extent of the sound image determined by the sound image information, at least one spread vector indicative of a position in the region, wherein the number of spread vectors is determined in advance. A gain calculation unit calculates, based on the at least one spread vector, a gain of each of audio signals supplied to two or more sound outputting units positioned in the proximity of the position indicated by the position information by using three-dimensional Vector Base Amplitude Panning.

A method for processing multi-channel audio signals, a system for processing multi-channel audio signals and a stereo apparatus comprising such a system. The multi-channel audio processing method comprises: receiving multi-channel audio signals from an external audio source, the multi-channel audio signals comprises a pair of surround channel signals and a pair of top channel signals; applying a crosstalk cancellation process in consideration of a head-related transfer function to the pair of top channel signals, so as to produce a pair of processed top channel signals, the head-related transfer function is configured to provide an elevation angle; mixing the pair of processed top channel signals, respectively, with the pair of surround channel signals, so as to produce a pair of mixed surround channel signals; providing the pair of mixed surround channel signals, respectively, to a pair of surround speakers.

Disclosed herein are system, apparatus, article of manufacture, method and / or computer program product embodiments, and / or combinations and sub-combinations thereof, for dynamically adjusting sound output of a multi-speaker configuration of a display device. The adjustment may be performed to optimize the sound output based on the physical position of the display device in relation to one or more microphones within a physical environment. An example embodiment operates by the speakers within the display device emitting calibration sound waves and receiving sound data associated with the calibration sound waves. The sound data includes characteristics of the calibration sound waves within the physical environment. The display device may then adjust sound output of speakers within the multi-speaker configuration based on the sound output characteristics.

A method for generating an audio signal associated with a virtual sound source is disclosed. The method comprises the steps of obtaining an input audio signal x(t) and modifying the input audio signal x(t) to obtain a modified audio signal. The latter step comprises performing a signal delay operation. Optionally, modifying the input audio signal comprises a signal inversion operation and / or a signal amplification or attenuation and / or a signal feedback operation. The method further comprises generating an audio signal y(t) based on a combination, e.g., a summation, of the input audio signal x(t) and the modified audio signal.

The present invention relates to an audio signal processing method, a parameterization device and an audio signal processing device for the same, and more particularly, to an audio signal processing method to implement filtering of an input audio signal with a low computational complexity, and a parameterization device and an audio signal processing device for the same. To this end, provided are a method for processing an audio signal, including: receiving an input audio signal; receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the truncated subband filter coefficients being filter coefficients of which energy compensation is performed based on the flag information and the length of at least one truncated subband filter coefficients being different from the length of the truncated subband filter coefficients of another subband; and filtering each subband signal of the input audio signal by using the truncated subband filter coefficients, and a parameterization device and an audio signal processing device for the same.