800results about "Two-channel systems" patented technology

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.

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.

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.

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.

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.

The invention relates to an audio processing method and device, a storage medium, a controller and a vehicle, and the method comprises the steps: receiving an audio signal inputted by a sound source, determining a processing mode of the audio signal according to the type of the sound source, processing the audio signal based on the processing mode, and obtaining a processed audio signal, and finally, the processed audio signal is transmitted to an audio playing assembly of the target sound area for playing, and the audio playing assembly comprises loudspeakers installed at at least two different target positions, so that a three-dimensional sound field space is constructed in the target sound area. Therefore, the audio processing effect can be improved on the basis of saving the cost, and an independent sound area and three-dimensional audio experience are provided for a user.

Techniques are described for wirelessly receiving, at first and second speaker assemblies, signals representing left and right audio channels from each of first and second sources, and using the signals, simultaneously playing, on both speaker assemblies, music, computer game audio, computer chat audio, and speech picked up by speaker microphones.

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 synthesizer (200) for synthesizing two-channel acoustic data from the single-channel acoustic data using the listener position or rotation; and a sound generator (300) for generating a two-channel audio signal from the audio signal and the two-channel acoustic data, where the two-channel synthesizer (200) is configured to separate (210) 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 (220, 230, 230) the two parts. 240) at least two portions to generate two-channel acoustic data for each portion, and wherein the two-channel synthesizer (200) is configured to calculate (230) the two-channel acoustic data for the early reflection portion using a specular portion describing different early reflections and a diffuse reflection portion describing diffuse reflection effects in the early reflection portion.

To provide a sound source generation device that makes it easier for a user to hear a target signal.SOLUTION: A sound source generation device includes estimation means for estimating sound source information of a sound source signal acquired by a sensor, generation means for generating playback information different from the sound source information and adding the playback information to a playback signal, and playback means for presenting the playback signal to a user, and the sound source information is the direction of arrival of the sound source signal, and the playback information is the direction of arrival of the playback signal.SELECTED DRAWING: Figure 1

Surround sound systems can dramatically expand the size of a user's sound field. Much surround sound content is mixed in a simplistic way where the front audio is copied to the rear, at a lower volume. It can be difficult for users to appreciate the value proposition of a surround sound system without more compelling spatially complex content. Quantifying surround sound complexity of various content items based on an audio spatial complexity scoring system can address this issue. Algorithms can be implemented to determine an audio spatial complexity score based on audio channels of a content item. Large catalog of content items can be analyzed, and audio spatial complexity scores can be associated with various content items. If a user has a surround sound system, content items with a high audio spatial complexity can be retrieved or recommended to the user to demonstrate the surround sound system's value better.

This signal processing device is provided with a detection circuit for detecting a temporal variation in a time difference between channels of a stereo signal, and a control circuit for controlling the degree of smoothing of an inter-channel cross correlation function on the basis of the temporal variation in the time difference between the channels.

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.

A method for speech equalization, comprising the steps of receiving an input audio signal, processing said input audio signal in dependence on frequency and to providing an equalized electric audio signal according to an equalization function, wherein said equalization function comprises at least an actuator part configured to dynamically applying a compensation filter to the received input signal and dynamically applying a transparent filter to the received input signal, and further transmitting an output signal perceivable by a user as sound representative of said electric acoustic input signal or a processed version thereof.

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, and wherein the two-channel synthesizer comprises two physically separate devices, wherein the first device of the two physically separated devices is configured to process at least one of the direct sound part and the early reflection part, wherein the second device of the two physically separated devices is configured to process at least one of the early reflection part and the late reverberation part, and wherein the first device and the second device are connected via a transmission interface and have separate power supplies.