1037results about "Two-channel systems" patented technology

Systems, devices, and methods for presenting audio associated with events associated with spatialized audio effects or non-spatialized audio effects in three-dimensional environments are disclosed. The spatialized and / or non-spatialized audio effects correspond with displaying, via a display generation component, a three-dimensional environment from a viewpoint of a user. While displaying the three-dimensional environment from the viewpoint of the user, the computer system detects an event. When the computer system detects an event which corresponds to a spatialized sound effect, the computer system presents the sound effect as emanating from a location in the three-dimensional environment associated with the event. When the computer system detects an event which corresponds to a non-spatialized sound effect, the computer system presents the sound effect as emanating from a predetermined location for audio in the three-dimensional environment that has a predetermined spatial relationship relative to the viewpoint of the user

A system for presenting audio content to a user. The system comprises one or more microphones coupled to a frame of a headset. The one or more microphones capture sound from a local area. The system further comprises an audio controller integrated into the headset and communicatively coupled to an in-ear device worn by a user. The audio controller identifies one or more sound sources in the local area based on the captured sound. The audio controller further determines a target sound source of the one or more sound sources and determines one or more filters to apply to a sound signal associated with the target sound source in the captured sound. The audio controller further generates an augmented sound signal by applying the one or more filters to the sound signal and provides the augmented sound signal to the in-ear device for presentation to a user.

A device and method for providing spatialized audio with dynamic head tracking that includes, in a static phase, providing a spatialized acoustic signal to a user that is perceived as originating from a virtual soundstage at a first location, and, upon determining one or more predetermined conditions are satisfied, which can include whether the users head has exceeded an angular bound, rotating the virtual soundstage to track the movement of the user's head.

A sound system capable of reducing the number of speakers required for performing localized reproduction in stereo is provided. A first speaker emits a sound based on an L (2n−1)-th processed acoustic signal with signal processing to make it audible in a first region in an L direction to an L (2n−1)-th acoustic signal and an R (2n)-th processed acoustic signal with signal processing to make it non-audible in a fourth region in an R direction to an R (2n)-th acoustic signal, and a second speaker emits a sound based on an L (2n)-th processed acoustic signal with signal processing to make it non-audible in a second region in an L direction to an L (2n)-th acoustic signal and an R (2n−1)-th processed acoustic signal with signal processing to make it audible in a third region in an R direction to an R (2n−1)-th acoustic signal.

The present application provides an audio processing method and apparatus, a wireless earphone, and a storage medium. A first wireless earphone receives a first to-be-presented audio signal sent by a playing device, and a second wireless earphone receives a second to-be-presented audio signal sent by the playing device; then, the first wireless earphone performs rendering processing on the first to-be-presented audio signal to obtain a first audio playing signal, and the second wireless earphone performs rendering processing on the second to-be-presented audio signal to obtain a second audio playing signal; and finally the first wireless earphone plays the first audio playing signal, and the second wireless earphone plays the second audio playing signal. Therefore, it is possible to achieve technical effects of greatly reducing the delay and improving the sound quality of the earphone since the wireless earphone can render the audio signals independently of the playing device.

An apparatus for encoding a multi-channel signal having at least two channels, has: a downmixer for calculating a downmix signal from the multi-channel signal; a parameter calculator for calculating a side gain from a first channel of the at least two channels and a second channel of the at least two channels and for calculating a residual gain from the first channel and the second channel; and an output interface for generating an output signal, the output signal having information on the downmix signal, and on the side gain and the residual gain.

Methods, apparatus, systems, and articles of manufacture are disclosed to generate spatial audio based on computer vision. An example apparatus includes at least one memory, instructions in the apparatus, and processor circuitry to execute the instructions to determine a position of an audio source based on an image generated via a camera, and apply an audio spatialization filter to an audio signal generated by a microphone based on the position of the audio source.

The present disclosure relates to a method and audio processing arrangement for extracting a target mid (and optionally a target side) audio signal from a stereo audio signal. The method comprises obtaining (S1) a plurality of consecutive time segments of the stereo audio signal and obtaining (S2), for each of a plurality of frequency bands of each time segment of the stereo audio signal, at least one of a target panning parameter (Θ) and a target phase difference parameter (Φ). The method further comprises extracting (S3), for each time segment and each frequency band, a partial mid signal representation (211, 212) based on at least one of the target panning parameter (Θ) and the target phase difference parameter (Φ) of each frequency band and forming (S4) the target mid audio signal (M) by combining the partial mid signal representations (211, 212) for each frequency band and time segment.

Systems and techniques are provided for performing spatial audio recording. For instance, a process can include detecting an occlusion for at least one audio frame of one or more audio frames associated with a spatial audio recording. During the spatial audio recording, the process can further include selecting, based on detection of the occlusion, at least one of an occluded spatial filter for the one or more audio frames or a non-occluded spatial filter for the one or more audio frames.

Disclosed is an audio codec that represents an immersive signal by a two-channel stereo signal that is a stereo rendering of the immersive signal and directional parameters. The directional parameters may be based on a perceptual model describing the direction of virtual speaker pairs to recreate the perceived location of dominant sounds. Audio processing at the decoder may be performed on the stereo signal in the frequency domain for multiple channel pairs using time-frequency tiles. Spatial localization of the audio signals may use a panning approach by applying weightings to the time-frequency tiles of the stereo signal for each output channel pair. The weightings for the time-frequency tiles may be derived based on the directional parameters, an analysis of the stereo signal, and the output channel layout. The weightings may be used to adaptively process the time-frequency tiles using a de-correlator to reduce or minimize spectral distortions from spatial rendering.

The present technology relates to an information processing device, an information processing method, and a program capable of accurately reproducing a reproduced sound in an acoustic space. The information processing device of the present technology includes: a harmonic signal generation unit configured to generate a first signal by convolving, to an input signal, transfer characteristics of harmonic distortion in an acoustic space; and a combining unit configured to combine together the first signal and a second signal, in which sound transmission characteristics excluding the harmonic distortion in the acoustic space is convolved to the input signal. The harmonic signal generation unit convolves, to the input signal respectively processed corresponding to the order the harmonic distortion, the transfer characteristics of the harmonic distortion for each order. The present technology can be applied to, for example, a system that performs audio mixing contents of a movie or the like.

An apparatus comprises a receiver (101) arranged to receive a frequency domain audio signal and a decorrelator (107) therefrom generates a frequency domain decorrelated signal. An envelope compensator (109) generates a compensated frequency domain decorrelated signal by applying a time domain envelope compensation to the decorrelated signal to reduce a difference between the time domain envelope of the frequency domain audio signal and the time domain envelope of the frequency domain decorrelated signal. The envelope compensator (109) comprises at least one trained artificial neural network having input values determined from at least one of the frequency domain audio signal and the frequency domain audio signal and generating time domain envelope compensation data for the time domain envelope compensation. An output generator (105) generates the multichannel audio signal by upmixing from the frequency domain audio signal and the frequency domain decorrelated signal.

Disclosed are an audio device and an intelligent wearable device. The audio device includes: a housing in which a first sound generation unit and a second sound generation unit are disposed, each of the first sound generation unit and the second sound generation unit has a front cavity and a rear cavity separated from each other, and the housing is provided with a first front sound hole in communication with the front cavity of the first sound generation unit, a second front sound hole in communication with the front cavity of the second sound generation unit, and a rear sound hole in communication with the rear cavity of the second sound generation unit. The audio device further includes a sliding baffle, wherein the sliding baffle is slidable between a first position and a second position with respect to the housing, and the sliding baffle is provided with a hole.

A speaker apparatus includes a pair of left and right speaker blocks that respectively have acoustic output sections and that are arranged inside a backrest. The pair of speaker blocks are arranged horizontally side by side relative to a headrest located on a front surface side of the backrest, and at least part of the headrest is located within a range of a nominal directivity angle of acoustics output from the acoustic output sections. This ensures that, of the acoustics output from the acoustic output section of the speaker block on the left side, at least some acoustics that head for the right ear are blocked by the headrest and that, of the acoustics output from the acoustic output section of the speaker block on the right side, at least some acoustics that head for the left ear are blocked by the headrest.

The invention discloses a panoramic sound effect control method, device and equipment. According to the method, a sound effect movement track used for indicating sound change characteristics can be constructed; on the basis of the sound effect motion trail, carrying out smoothing processing on the motion speed and angle parameters of a target sound source; determining all playing sound channels, and generating an acoustic image optimization parameter corresponding to each playing sound channel; and optimizing the smoothed target sound source based on the sound image optimization parameter of each playing sound channel, and rendering the optimized corresponding target sound source in each playing sound channel. Visibly, according to the method and the device, the problems of sound image breakage and speed mismatch distortion in the motion simulation process can be avoided by accurately grasping the sound change characteristics, optimizing the sound motion parameters and individually regulating and controlling each playing sound channel, the natural fluency of the playing effect of the target sound source is comprehensively improved, and the user experience is improved. And the playing quality of the target sound source and the auditory experience of audiences are comprehensively improved.

A display method of the present disclosure includes a first acquisition step of acquiring, based on first sound data having a first bit depth generated based on a sound signal output from a first sound collection device, second sound data having a second bit depth larger than the first bit depth, and an output step of outputting, to a display device, a volume video representing a volume related to the second sound data.

A method comprises: obtaining softmask values for frequency bins of time-frequency tiles representing an audio signal; reducing, or expanding and limiting, the softmask values; and applying the reduced, or expanded and limited, softmask values to the frequency bins to create a time-frequency representation of an estimated target source. An alternative method comprises, for each time-frequency tile: obtaining softmask values; applying the softmask values to the frequency bins to create a time-frequency domain representation of an estimated target source; obtaining a panning parameter and a source concentration estimates for the target source; determining, using the panning parameter estimate and the softmask values, a magnitude for the time-frequency representation of the estimated target source; determining, using the panning parameter estimate and the source phase concentration estimate, a phase for the time-frequency representation of the estimated target source; and combining the magnitude and the phase.

An apparatus includes a processor configured to perform signal enhancement of an input audio signal to generate an enhanced monaural audio signal. The processor is further configured to mix the first audio signal and the second audio signal to generate a stereo audio signal. The first audio signal is based on the enhanced monaural audio signal.

A method for audio signal filtering. The method includes generating a pair of filters for a certain location specified by an elevation angle ϑ and an azimuth angle φ, the pair of filters consisting of a right filter (ĥr(ϑ, φ)) and a left filter (ĥl(ϑ, φ)); filtering an audio signal using the right filter; and filtering the audio signal using the left filter. Generating the pair of filters comprises: i) obtaining at least a first set of elevation basis function values at the elevation angle; ii) obtaining at least a first set of azimuth basis function values at the azimuth angle; iii) generating the right filter using: a) at least the first set of elevation basis function values, b) at least the first set of azimuth basis function values, and c) right filter model parameters; and iv) generating the left filter using: a) at least the first set of elevation basis function values, b) at least the first set of azimuth basis function values, and c) left filter model parameters.

In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Audio signal processor for generating a two-channel audio signal has: 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, and wherein the two-channel synthesizer is configured to calculate the two-channel acoustic data for the early reflection part using a specular part describing distinct early reflections and a diffuse part describing a diffuse influence in the early reflection part.

A head-mounted device for generating binaural audio content includes a first rod coupled to a first microphone housing enclosing a first front microphone and a first rear microphone, respectively, that generate acoustic signals. The first microphone housing includes a first front port facing downward and a first rear port facing rearward. The apparatus includes a second rod coupled to a second microphone housing enclosing a second front microphone and a second rear microphone, respectively, that generate acoustic signals. The second microphone housing includes a second front port facing downward and a second rear port facing rearward. In one embodiment, an apparatus includes a binaural audio processor including a beamformer and a storage device. The beamformer generates a first beamformer signal based on acoustic signals from the first front microphone and the first rear microphone and a second beamformer signal based on acoustic signals from the second front microphone and the second rear microphone. A storage device stores the first and second beamformer signals as a two-channel file.

Disclosed are systems, apparatuses, processes, and computer-readable media to. According to some aspects, a method of processing audio data may include obtaining, at a computing device, sensing information from an audio device outputting a spatial audio stream for a user, wherein the audio device includes a first audio output device and a second audio output device; determining, based on the sensing information, that the second audio output device is not in use; modifying the spatial audio stream based on determining that the second audio output device is not in use and a head pose of the user to create a modified spatial audio stream; and providing the modified spatial audio stream to the first audio output device.

System and method for enhancing audio reproduced by an audio reproduction device is disclosed. A spatial emulation appliance with a processor and memory is provided. Media content for reproduction is received by the spatial emulation appliance. A selection of a physical space to render the reproduction of the media content is received by the spatial emulation appliance. Convolved media content is generated by the spatial emulation appliance using filter coefficients, rendering logic and other parameters representative of the selected physical space. The convolved media content is presented for reproduction. A non-fungible token (NFT) is generated indicative of the authenticity and the spatial characteristics of the selected physical space.

A method performed by a programmed processor of an audio system, the method includes retrieving a head-related transfer function (HRTF) dataset from memory, processing the HRTF dataset to remote comb filtering effects during audio playback; applying, to each HRTF in the processed HRTF dataset, a gain to increase a magnitude of the HRTF at a frequency range; and storing the processed HRTF dataset in memory.

The invention discloses an audio device and a control method thereof, and a storage medium, and relates to the technical field of audio, and the method comprises the steps: monitoring the head rotation angle of a target user, and obtaining the personalized acoustic transmission data of the target user, the personalized acoustic transmission data comprises at least one head related transfer function HRTF obtained by training based on the personal auditory physiological data of the target user; according to the head rotation angle, adjusting a pose of a virtual loudspeaker in the audio equipment to obtain a virtual playing pose of the virtual loudspeaker; and acquiring a target head-related transfer function HRTF corresponding to the virtual playing pose in the personalized acoustic transmission data, and rendering audio played by the audio equipment according to the target head-related transfer function HRTF so as to simulate audio playing at the virtual playing pose. According to the invention, the problem of poor sound quality performance of the audio equipment caused by head rotation is solved.