255results about "Pseudo-stereo systems" patented technology

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 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.

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).

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

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

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

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

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.

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.

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

Techniques for shaping sound based on a speaker orientation are disclosed. In an embodiment, an audio data stream is obtained by a zone player having one or more speaker drivers, an orientation of the zone player is determined, and sound is reproduced by the zone player based on the orientation. The sound may be further shaped based on other states of the zone player in addition to orientation. The overall sound may be shaped from one zone player or from a collection of zone players.

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

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

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

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

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 input interface (100) is configured to acquire (150) an original representation related to the single-channel acoustic data, and deriving (151) single-channel acoustic data using the original representation and additional data stored in or accessible by the audio signal processor.

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

Disclosed are an audio upmixing method and an audio apparatus. The method includes: obtaining (202) a stereophonic audio signal, and performing feature extraction on the stereophonic audio signal to obtain a stereophonic audio feature; extracting (204) a plurality of left channel audio signals and a plurality of right channel audio signals from the stereophonic audio feature based on a plurality of audio output channels, where the plurality of audio output channels are independent of each other, and each of the plurality of audio output channels is configured to output the corresponding left channel audio signal or the corresponding right channel audio signal; fusing (206) the left channel audio signal and the right channel audio signal corresponding to one target channel to obtain first audio signals of a plurality of target channels, where each of the plurality of target channels corresponds to two of the plurality of audio output channels; and outputting (208) an audio upmixing signal of a target format based on the first audio signals of the plurality of target channels, where the target format corresponds to the plurality of target channels. An audio upmixing signal of a target format is output based on the first audio signals of the plurality of target channels, where the target format corresponds to the plurality of target channels. This method can improve an audio upmixing effect. (Fig. 1)

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 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, wherein the two-channel synthesizer (200) is configured to segment (231) the early reflection part into a plurality of segments, to determine (232) a plurality of image source positions representing source positions of reflection sound, to associate the image source positions to the segments using a matching operation, wherein the matching operation comprises calculating a time of sound arrival for each image source to the listener position and associating (234) the image source positions to corresponding segments that have time delays in the corresponding segments best matching with the time of sound arrival of the corresponding image source positions, and to calculate the two-channel acoustic data for the direct sound using the image source positions associated to the segments.

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

A method of audio processing includes performing object extraction on a binaural audio signal to generate a principal component signal and a residual component signal. The system can process the principal and residual components using different processing parameters to generate a processed binaural signal that provides an improved listening experience.

The invention relates to an audio upmixing method and audio equipment. The method comprises the following steps: obtaining a stereo audio signal, and carrying out feature extraction on the stereo audio signal to obtain stereo audio features; according to a plurality of audio output channels, a plurality of left channel audio signals and a plurality of right channel audio signals are extracted from the stereo audio features, the plurality of audio output channels are mutually independent, and each audio output channel is used for outputting the corresponding left channel audio signal or the corresponding right channel audio signal; the left channel audio signals and the right channel audio signals corresponding to the same target channel are fused to obtain first audio signals of a plurality of target channels, and each target channel corresponds to two audio output channels; and according to the first audio signals of the plurality of target sound channels, outputting an audio upmix signal in a target format, the target format corresponding to the plurality of target sound channels. By adopting the method, the audio upmixing effect can be improved.