400results about "Pseudo-stereo systems" patented technology

The invention relates to the field of audio playing control, in particular to an audio playing control method and device, equipment and a storage medium. The method comprises the following steps: acquiring a real-time playing audio signal and a spatial multi-angle reflection audio signal; performing one-by-one angle signal reflection time calculation and audio spatial three-dimensional form modeling on the spatial multi-angle reflection audio signals to construct a three-dimensional audio spatial form model; performing spatial angle reflection attenuation evolution on the spatial multi-angle reflection audio signals one by one, and performing spatial audio attenuation correction based on the three-dimensional audio spatial form model so as to obtain spatial attenuation compensation audio signals; and carrying out audio rhythm fluctuation analysis on the spatial attenuation compensation audio signal, and carrying out deep audio semantic analysis so as to obtain an audio semantic feature of the audio signal. According to the invention, personalized audio playing is realized, the audio parameters are dynamically adjusted based on the environmental noise, and the audio experience of the user is improved.

The invention provides an up-mixing processing method, system and device for a stereo audio video, and a storage medium, and relates to the technical field of audio and video processing, and the method comprises the steps: obtaining a to-be-processed stereo audio video, carrying out the multi-stream separation of the to-be-processed stereo audio video, obtaining a plurality of independent sound source components, the to-be-processed stereo audio video comprises a to-be-processed stereo audio or a to-be-processed stereo video; performing dry and wet sound processing on the to-be-processed stereo sound video to obtain an environment sound component; performing frequency band extension on the environment sound component to obtain a target environment sound component; and carrying out space rendering on each independent sound source component and the target environment sound component by adopting a vector-based amplitude translation method, and carrying out multichannel output fusion to obtain a multichannel audio / video. According to the method, the sound source is accurately positioned and efficiently separated, the dynamic sense and the space sense of the sound source object are enhanced, and the audio and video upmix processing quality and the user experience are improved.

An audio upmixing method and an audio apparatus are disclosed. The method comprises: performing feature extraction on a stereophonic audio signal to obtain a stereophonic audio feature; and extracting channel audio signals and right channel audio signals from the stereophonic audio feature based on audio output channels. The audio output channels are independent of each other, and each audio output channel is configured to output the corresponding left channel audio signal or the corresponding right channel audio signal. The method further comprises fusing the left channel audio signal and the right channel audio signal corresponding to a target channel to obtain first audio signals of a plurality of target channels. Each target channel corresponds to two audio output channels. The method further comprises outputting an audio upmixing signal of a target format based on the first audio signals. The target format corresponds to the plurality of target channels.

A sound control system includes a sound generating device disposed at a vehicle ceiling corresponding to a seat region of the vehicle and at a sound space formed within the vehicle to correspond to the seat region, and a sound processing circuit for providing a vibration driving signal to the sound generating device. The sound generating device vibrates an interior material of the vehicle ceiling based on the vibration driving signal.

A filter system comprises a filter converter (101) and a filter compressor (102) for generating compressed subband filter impulse responses from input subband filter impulse responses corresponding to subbands, which comprise filter impulse response values at filter taps. The filter compressor comprises a processor (820) for examining the filter impulse response values from at least two input subband filter input responses to find filter impulse response values having higher values and at least one filter impulse response value having a value being lower than the higher values, and a filter impulse response constructor (305) for constructing the compressed subband filter impulse responses using the filter impulse response values having the higher values, wherein the compressed subband filter impulse responses do not include filter impulse response values corresponding to filter taps of the at least one filter impulse response value having the lower value or comprise zero-valued values corresponding to filter taps of the at least one filter impulse response value having the lower value.

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.

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.

Real-time source separation is desirable for its ability to clearly transmit desired audio signals but is computationally costly when applied across large areas. Thus, a tradeoff between sound source separation quality and system performance is presented. For that reason, this disclosure of area-based sound source separating techniques resolves this tradeoff. Concepts herein include an area-based sound source separating machine learning architecture defines subareas and redefines those subareas as necessary to capture desired audio signal(s). By redefining the subarea, less noise or unwanted audio signals are received / processed, significantly reducing computational resources while increasing throughput of the desired audio signal(s).

An apparatus for assisting spatial rendering in at least two acoustic environments, the apparatus including circuitry configured to: obtain a listener position; determine a first acoustic environment of the at least two acoustic environments based on the listener position; determine at least one second acoustic environment of the at least two acoustic environments based on the first acoustic environment, wherein the second acoustic environment is coupled to the first acoustic environment by an active portal connection; obtain at least one second acoustic environment input audio signal; and generate at least one listener output audio signal based at least partly on the application of processing of the at least one second acoustic environment input audio signal by at least one second acoustic environment reverberator and then a first acoustic environment reverberator.

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.

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.

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 method, system, and medium for processing audio are provided. In some embodiments, a method for processing audio includes receiving a stereo audio signal. The method may include separating the stereo audio signal into a steer signal and a diffusion signal. The method may include determining one or more diffusion signal modification parameters based on a current listening context, where the one or more diffusion signal modification parameters indicate a proportion of the diffusion signal to be redistributed to one or more output channels in an output multi-channel signal, or a degree of attenuation to be applied to the diffusion signal. The method may include generating the output multi-channel signal based on the steer signal, the diffusion signal, and the one or more diffusion signal modification parameters. The method may include providing the output multi-channel signal to a virtualizer for rendering as a binaural audio signal for playback on a wearable device.

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.

Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.

An audio signal processor for generating a two-channel audio signal, comprises: 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 a listener position or rotation; and a sound generator (300) for generating the two-channel audio signal from an audio signal and the two-channel acoustic data, wherein 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, 240) the at least two parts for generating two-channel acoustic data for each part, to determine (222), for the listener position and a source position or orientation of a sound source, directivity information of the sound source, and to use the directivity information in the calculation (220) of the two-channel acoustic data for the direct sound part.

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.

To reduce processing load on a receiving side when transmitting multiple types of audio data.SOLUTION: A meta file containing meta information for acquiring a predetermined number of audio streams, each containing encoded data from multiple groups, by a receiving device is transmitted. Attribute information indicating respective attributes of the encoded data of the multiple groups is inserted into this meta file. For example, stream correspondence information, which indicates which audio stream includes each of the encoded data of the multiple group, is further inserted into the meta file.SELECTED DRAWING: Figure 17

The present disclosure relates to systems, methods, and devices for audio reproduction. The system includes a media computing device and an electronic device that are connected so that their speakers can be used in combination when optimizing audio reproduction of media being played or streamed. Each device can be portable, and the audio reproduction can include virtual / surround / soundfield audio reproduction. When connected, the media computing device functions as the host for data communication, and the electronic device functions as the slave. In this way, the media computing device can process audio signals into any parts for reproduction on any speakers. Additionally, when connected and if needed, the electronic device functions as the host for power management, and the media computing device functions as the slave. Depending on the system configuration, the appropriate connection can be made via a wireless connection or a single-cable wired connection between the media computing device and the electronic device.

Audio objects are associated with positional metadata. A received downmix signal comprises downmix channels that are linear combinations of one or more audio objects and are associated with respective positional locators.In a first aspect, the downmix signal, the positional metadata and frequency-dependent object gains are received. An audio object is reconstructed by applying the object gain to an upmix of the downmix signal in accordance with coefficients based on the positional metadata and the positional locators.In a second aspect, audio objects have been encoded together with at least one bed channel positioned at a positional locator of a corresponding downmix channel. The decoding system receives the downmix signal and the positional metadata of the audio objects. A bed channel is reconstructed by suppressing the content representing audio objects from the corresponding downmix channel on the basis of the positional locator of the corresponding downmix channel.

An audio system comprises a processing unit, and a reverb classification unit, wherein the reverb classification unit is configured to receive a first plurality of audio input signals, estimate a class of reverberation suitable for the first plurality of audio input signals using a deep learning (DL) classification algorithm, and output a prediction to the processing unit, the prediction including information concerning the estimated class of reverberation, and the processing unit is configured to receive the first plurality of audio input signals, generate a second plurality of audio output signals based on the first plurality of audio input signals, and output the second plurality of audio output signals, wherein generating the second plurality of audio output signals comprises adding reverberation to at least one of the second plurality of audio output signals based on the prediction received from the reverb classification unit.

This application discloses a delay estimation method and apparatus, and belongs to the audio processing field. The 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, so as to resolve a problem that the cross-correlation coefficient is excessively smoothed or insufficiently smoothed, thereby improving accuracy of estimating an inter-channel time difference.

The method of sound processing according to the invention takes into account the quantum effect of the propagation of the sound in the material from which the theatre walls are built, what allows to reconstruct the timbre of the processed sound. The sound processed with the method according to the invention is clearer, deeper and stronger than the original. The method consists in the digital filtering of the sound with the aid of finite impulse response obtained from pairs of signals delay-attenuation computed for the trajectories originating from the sound source and composed from the trajectory segments in the air and in the material of theatre walls. The method thereof is characterized by the fact that the first directions of the trajectory segments in the air are selected randomly and the next ones are determined according to the rule that the reflection angle from the theatre wall is the same as the incidence angle till the moment when the trajectory reaches the listener's location. In the advantageous embodiments of the invention the material from which the theatre walls are built is wood with the density 400 kg / m3 and with the sound propagation speed 4000 m / s, the theatre building is equipped with the stage and the wall behind the stage, the source of sound propagation are the actor's lips where the mentioned actor is on the stage or in the middle of orchestra and the listener sits in the middle of the first row.

Methods and systems for advanced stereo processing of an audio signal are disclosed. The methods and systems include selecting a coding mode of either transform coding or linear predictive coding and performing advanced stereo processing when in the selected coding mode. Both encoding and decoding operations are provided.

There is provided an apparatus for rendering an audio scene to be consumed by a listener in a physical space, the apparatus including circuitry configured to: obtain at least one information to render the audio scene with at least one audio signal according to the at least one information; obtain play-area information, the play-area information including information about a region within the physical space and within which the listener is allowed to physically move within, the play-area information including at least one of: geometry; position; orientation; and coordinate system; generate an audio rendering using the at least one audio signal in accordance with the play-area information; and obtain a rendering control parameter based on the play-area information, wherein the rendering control parameter is configured to update the audio rendering.

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

A multi-input, multi-output audio processing is implemented as a linear system for use in an audio filter bank to convert a set of frequency-domain input audio signals into a set of frequency-domain output audio signals. The transfer function from one input to one output is defined as a frequency-dependent gain function. In some implementations, the transfer function includes a direct component that is substantially defined as a frequency-dependent gain and one or more decorrelating components having a frequency-varying group phase response. The transfer function is formed by a set of sub-band functions, where each sub-band function is formed by a set of corresponding component transfer functions that include the direct component and one or more decorrelating components.