2134results about "Stereophonic systems" patented technology

The invention relates to the technical field of audio processing, in particular to a self-adaptive scene sound effect adjusting method and device and a storage medium, and the method comprises the steps: collecting a mixed audio stream and an original playing audio, and extracting an acoustic fingerprint vector through a pre-trained convolutional neural network; performing acoustic feature deconstruction on the acoustic fingerprint vector to obtain a plurality of acoustic feature components, performing scene recognition judgment according to a preset scene judgment rule based on the plurality of acoustic feature components, and determining an audio playing scene; constructing a time-frequency mask according to the audio playing scene, generating a noise reduction gain matrix based on the time-frequency mask, and performing scene sound field enhancement on the audio playing scene to obtain a sound field optimization matrix; and performing noise reduction and sound effect adjustment on the original playing audio according to the noise reduction gain matrix and the sound field optimization matrix, and outputting the optimized playing audio. According to the invention, sound effect adjustment can be dynamically adapted according to different scenes, and the noise reduction accuracy and the multi-scene sound quality adaptability are improved.

Methods and audio processing units for generating an object based audio program including conditional rendering metadata corresponding to at least one object channel of the program, where the conditional rendering metadata is indicative of at least one rendering constraint, based on playback speaker array configuration, which applies to each corresponding object channel, and methods for rendering audio content determined by such a program, including by rendering content of at least one audio channel of the program in a manner compliant with each applicable rendering constraint in response to at least some of the conditional rendering metadata. Rendering of a selected mix of content of the program may provide an immersive experience.

In aspects of adjusting audio output in a vehicle, a vehicle audio system implements an audio playback manager that detects a location of a first person in the vehicle, the first person located in proximity of a first speaker device configured for audio output. The audio playback manager also detects an additional location of a second person in the vehicle, the second person located in proximity of a second speaker device configured for the audio output. The audio playback manager determines a preference for the first person related to the audio output and an additional preference for the second person related to the audio output. Based on the preference of the first person or the additional preference of the second person, the audio playback manager adjusts a volume of at least one of the first speaker device or the second speaker device.

A computer-implemented sound calibration method, a system and a storage medium are provided. A test audio is played by a subwoofer and a main speaker, transmitted audio is collected by an audio collection device at a primary listening position, a sound generation delay and a volume parameter difference are analyzed, and the subwoofer is calibrated based on of a frequency response curve of a scenario, so as to solve the problem in the related art where the user experience is adversely affected due to low sound calibration accuracy between the subwoofer and the main speaker.

An augmented reality display system comprises a passable world model data comprises a set of map points corresponding to one or more objects of the real world. The augmented reality system also comprises a processor to communicate with one or more individual augmented reality display systems to pass a portion of the passable world model data to the one or more individual augmented reality display systems, wherein the piece of the passable world model data is passed based at least in part on respective locations corresponding to the one or more individual augmented reality display systems.

On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (α) from the bitstream, where 1 ≤ m < n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

A system for controlling a plurality of lighting devices (11-15) to render light effects accompanying a rendering of audiovisual content (81) is configured to associate, based on a first characteristic of a first audio channel / object (86), the first audio channel / object with a first lighting device (12) and not with a second lighting device (13). The first characteristic is indicative of an audio source position. The system is further configured to associate a second audio channel (87) with the first and second lighting devices. The system is further configured to determine whether second audio content in the second audio channel meets predetermined criteria and determine a first light effect based on a determined chromaticity. If the one or more predetermined criteria are not met, the light intensity is based on first audio content in the first audio channel / object, and otherwise, the light intensity is based on the second audio content.

The invention discloses a college dormitory noise multi-dimensional evaluation method based on psychoacoustic and sound field simulation, and belongs to the field of building environment and noise control. The objective of the invention is to solve the problem that an existing dormitory noise evaluation method is single in index and neglects subjective perception and spatial distribution heterogeneity. The method comprises the following steps: acquiring physical data and original audio of dormitory noise through multi-point synchronous acquisition; calculating loudness, sharpness, roughness, fluctuation intensity and comprehensive psychoacoustic annoyance (PA) by using a psychoacoustic model, and scientifically quantifying subjective interference of noise; meanwhile, sound field simulation is carried out based on the accurately established three-dimensional dormitory model, a sound pressure level distribution cloud picture is generated, and a standing wave phenomenon and a sound energy accumulation area are visually revealed; and finally, generating a targeted noise reduction optimization strategy by integrating subjective and objective and spatial analysis results. According to the method, full-chain diagnosis of physical characteristics-subjective perception-spatial distribution of dormitory noise is realized, and a scientific basis is provided for accurate and efficient noise treatment.

In some embodiments a computer-implemented method may include determining a position of a device relative to an environment, generating spatial audio content that provides natural language contextual descriptions of the environment based on the determined position of the device and presenting the spatial audio content to a user, wherein the spatial audio content dynamically updates in response to changes in the position of the device or the environment.

The present disclosure provides an audio system, method, and system for facilitating machine operation. The machine includes an actuator that assists the tool in performing the task. In an example, an audio system is configured to receive an audio mixed signal of a signal generated by an audio source that includes at least one of a tool or an actuator that is performing a task. An audio source forming an audio mixed signal is identified by a relative position to each microphone of a microphone array that measures the audio mixed signal. The audio system is configured to extract an audio signal from an audio mixed signal generated by the identified audio source based on a correlation of spectral features in a multi-channel spectrogram of the audio mixed signal and directional information indicative of a relative position of the identified audio source. The audio system outputs the extracted audio signal to facilitate operation of the machine.

Methods and systems for generating a sound. The method includes obtaining meshes representing at least one object in a frame of an environment, the environment including a source and a receiver; determining spatial continuity information and reflection normal information of the meshes; determining, based on the spatial continuity information and the reflection normal information, reflection paths between the source and the receiver involving the at least one object, each of the reflection paths having at least one reflection point associated with the at least one object; obtaining spatially sampled results by spatially sampling a space around the at least one reflection point of reflection paths using multiple distributions of rays, the spatially sampled results correlating with geometric information of the meshes; generating reflection amplitude responses for each of multiple audible frequencies in the environment based on the spatially sampled results; and producing a sound based on the reflection amplitude responses.

A loudspeaker system includes a three-loudspeaker array and signal processing method for rendering and playing audio content in a listening environment. The three-loudspeaker array includes a left loudspeaker element, a center / height loudspeaker element, and a right loudspeaker element. A signal processor / renderer can be programmed or configured to process input audio content with a plurality of (e.g., at least five) playback channels (e.g., left, center, right, left elevation, and right elevation channels). The right loudspeaker can be driven by a right main signal and a right elevation-cancellation signal, and / or a combined signal thereof. The left loudspeaker can be driven by a left main signal and a left elevation-cancellation signal, and / or a combined signal thereof. The center / height loudspeaker can be driven by a combination of a center main signal and an elevation / height signal derived or rendered from the input audio signal.

In some embodiments, a method comprises executing, by a head-tracked audio system, a first algorithm configured to dynamically update a forward head angle and a spatial orientation in response to a detected head movement, and a second algorithm configured to adjust a recentering filter based on the detected head movements, and maintaining, by the head-tracked audio system through execution of the first algorithm and the second algorithm, a spatial audio placement relative to a wearer during the detected head movements.

For example, one object is to realize more excellent sound field correction. An information processing apparatus includes a control unit that converts a transfer characteristic of a sound pressure from a speaker device to a listening position into an energy characteristic, and calculates a correction parameter for correcting a sound field generated by an output sound of the speaker device using the converted energy characteristic.

Systems can include an acoustic sensor array configured to receive acoustic signals, an illuminator configured to emit electromagnetic radiation, an electromagnetic imaging tool configured to receive electromagnetic radiation, a distance measuring tool, and a processor. The processor can illuminate the target scene via the illuminator, receive electromagnetic image data from the electromagnetic imaging tool representative of the illuminated scene, receive acoustic data from the acoustic sensor array, and receive distance information from the distance measuring tool. The processor can be further configured to generate acoustic image data of the scene based on the received acoustic data and received distance information and generate a display image comprising combined acoustic image data and electromagnetic image data. The processor can determine depths of various acoustic signals within a scene and generate a representation of the scene the shows the determined depths, including floorplan and volumetric representations.

Example embodiments relate to an apparatus, method and computer program for output of stereo or spatial audio. The method may for example comprise providing first data representing stereo or spatial audio, providing second data representing other audio and determining an orientation and / or configuration of an apparatus. The method may also comprise outputting the first data to selected first and second loudspeakers of the apparatus, comprising at least three loudspeakers based, at least in part, on the orientation and / or configuration of the apparatus; and outputting the second data to at least a third loudspeaker of the at least three loudspeakers other than the first and second loudspeakers.

The invention relates to the technical field of sound correction, and discloses a room sound calibration method based on listening tendency control and related equipment, which are used for improving the matching degree of a calibration result and listening preference of a user. The room sound calibration method based on listening tendency control comprises the following steps: obtaining a listening tendency parameter selected by a user through an interactive interface and a room pulse response of a current room; determining a group of frequency-related window control parameters according to the hearing tendency parameter and the room pulse response; determining a windowing strategy with the window length changing along with the frequency based on the window control parameters; windowing the room impulse response by using a windowing strategy to extract an effective frequency response curve for acoustic calibration; calculating according to the difference between the effective frequency response curve and the target frequency response curve to obtain a correction filter coefficient; and calibrating the audio signal of the audio playing system based on the correction filter coefficient.

A mixing matrix, suitable for converting a scene-based audio (SBA) input signal to an object-based audio (OBA) signal, is constructed so that the resulting OBA signal is composed of object signals with amplitudes that are biased according to amplitude preference coefficients. The amplitude preference coefficients are chosen to place dominant spatial audio objects in a fewer number of output object channels, to provide a more discrete OBA rendering of the SBA input signal.

An apparatus for rendering spatial audio with reverberation receive a spatial audio stream, the spatial audio stream comprising at least one audio signal and spatial metadata associated with the at least one audio signal The apparatus further obtains at least one room effect parameter comprising at least one reverberation time, adjusts at least one of the at least one room effect parameter when the at least one reverberation time is less than a determined threshold value and generates a spatial audio signal using the spatial audio stream and the adjusted at least one room effect parameter, the generated spatial audio signal comprising a first spatial audio portion and processed dependent on the spatial metadata and a second spatial audio portion generated based on the at least one audio signal and processed dependent on the adjusted at least one room effect parameter.

An audiovisual system uses a spatial position detection module to process audio signals and determine the location and orientation of a speaking participant. Based on this data, the system dynamically controls sensors to optimize audio and video capture. Behavioral and contextual information may also be used to train intelligence models for improved system performance. Further, sensor arrays may be utilized to identify gaze vectors of participants to select a camera sensor of the sensor array. Meeting content collected with the sensor array may be organized into meeting metrics in accordance with an analytics strategy before training an intelligence module with the meeting metrics. Meeting content may also be configured into digital tiles in accordance with a tile strategy. At least one of the digital tiles may be altered in response to a meeting condition detected by the sensor array.

The present disclosure relates to a method and an apparatus for adding reverberation, a device, and a product. The method includes: separating reverberant audio based on wet audio, where the wet audio includes dry audio and the reverberant audio corresponding to the dry audio. The method further includes: determining, based on the reverberant audio, reverberation parameters for indicating reverberation effects of the reverberant audio. In addition, the method further includes: adding reverberation to synthesized dry audio based on the reverberation parameters.

Methods are provided of three dimensionally synthetizing a digital audio signal depending on spatial commands and filter instructions. Such methods comprise: performing, depending on the spatial commands, a spatialization method at audio rate on the digital audio signal to spatialize it and therefore produce a spatialized audio signal whose reproduction implements, in a three dimensional, 3D, audible space, spatialized virtual sound sources defined by spatial parameters; performing, depending on the filter instructions, digital filtering on the spatialized audio signal to produce an evolved or synthetised audio signal whose reproduction implements, in the 3D audible space, evolved virtual sound sources; and providing the evolved audio signal to a plurality of loudspeakers or headphones to reproduce the evolved audio signal and, therefore, implement the evolved virtual sound sources in the 3D audible space. Systems, computer systems and computer programs that are suitable for performing such methods are also provided.

Some systems include an acoustic sensor array configured to receive acoustic signals, an electromagnetic imaging tool configured to receive electromagnetic radiation, a user interface, a display, and a processor. The processor can receive electromagnetic data from the electromagnetic imaging tool and acoustic data from the acoustic sensor array. The processor can generate acoustic image data of the scene based on the received acoustic data, generate a display image comprising combined acoustic image data and electromagnetic image data, and present the display image on the display. The processor can receive an annotation input from the user interface and update the display image based on the received annotation input. The processor can be configured to determine one or more acoustic parameters associated with the received acoustic signal and determine a criticality associated with the acoustic signal. A user can annotated the display image with determined criticality information or other determined information.

The invention relates to the technical field of audio sound calibration, and discloses a room sound calibration method based on intermediate frequency reverberation time and related equipment, which are used for improving the accuracy of a sound calibration result. The room sound calibration method based on the intermediate frequency reverberation time comprises the following steps: acquiring a room impulse response of a target room and calculating the intermediate frequency reverberation time; according to the intermediate frequency reverberation time, the recommended window length of the frequency correlation window is determined by querying a predefined mapping relation, and the mapping relation is established based on acoustic association between the room reverberation time and the analysis window length; based on the recommended window length, determining the actual window length of the actual application in combination with the user setting mode; performing time domain windowing and frequency-related transfer function estimation processing on the room impulse response according to the actual window length to obtain window function weighted transfer function estimation; filter calibration coefficients are generated based on the transfer function estimates and applied for audio calibration.

The embodiment of the invention provides a method executed by electronic equipment, the electronic equipment and a storage medium, and relates to the field of artificial intelligence. The method comprises the following steps: acquiring an input first audio signal; based on direction of arrival (DOA) information of a first audio signal, target direction information and spatial information of a physical microphone (RM) of the electronic equipment, obtaining the number and positions of virtual microphones (VM); a second audio signal is obtained based on the first audio signal and the number and location of VMs. Optionally, the method performed by the electronic device may be performed using an artificial intelligence model.

The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The method comprises performing frequency-domain modifications selectively before or after upmixing.

This application discloses a method and apparatus for processing three-dimensional audio signals, used to implement bit allocation of the signal. The method includes: spatially encoding the three-dimensional audio signal to be encoded to obtain a transmission channel signal and transmission channel attribute information, wherein the transmission channel signal includes: at least one virtual speaker signal group and at least one residual signal group; and determining the bit allocation ratio of the virtual speaker signal group and the bit allocation ratio of the residual signal group based on the transmission channel attribute information.

A method performed by an audio system that includes a headset for playing back audio, the method includes receiving an audio signal having audio content, determining a characteristic of the audio content, determining a parameter of the headset, producing a filtered audio signal by applying an equalization (EQ) filter to the audio signal based on the characteristic of the audio content and the parameter of the headset, and driving one or more speakers of the headset with the filtered audio signal.

This application provides a sound-field control method and device, an electronic device and a computer-readable storage medium. The method includes: obtaining a target sound field parameter, calculating a control parameter of a sound-generating device according to the target sound field parameter, and controlling operation of the sound-generating device according to the control parameter.

The invention provides methods and devices for stereo encoding and decoding using complex prediction in the frequency domain. In one embodiment, a decoding method, for obtaining an output stereo signal from an input stereo signal encoded by complex prediction coding and comprising first frequency-domain representations of two input channels, comprises the upmixing steps of: (i) computing a second frequency-domain representation of a first input channel; and (ii) computing an output channel on the basis of the first and second frequency-domain representations of the first input channel, the first frequency-domain representation of the second input channel and a complex prediction coefficient. The method comprises applying independent bandwidth limits for the input channels.