56results about "Two-channel systems" patented technology

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.

The present invention provides a method for processing music audio data, comprising the steps of providing input audio data representing a first piece of music containing a mixture of predetermined musical timbres, decomposing the input audio data to generate at least a first audio track representing a first musical timbre selected from the predetermined musical timbres, and a second audio track representing a second musical timbre selected from the predetermined musical timbres, applying a predetermined first audio effect to the first audio track, applying no audio effect or a predetermined second audio effect, which is different from the first audio effect, to the second audio track, and obtaining recombined audio data by recombining the first audio track with the second audio track.

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.

This disclosure describes the classification (hereinafter referred to as "UNCLR classification") and crosstalk detection (hereinafter referred to as "XTALK detection") of irrelevant stereo content in an input stereo audio signal. This disclosure also describes stereo mode selection, such as automatic LRTD / DFT stereo mode selection. Furthermore, this disclosure uses the classification to select one of a first stereo mode and a second stereo mode for encoding a stereo audio signal including the left and right channels; to detect crosstalk in the stereo audio signal including the left and right channels in response to features extracted from the stereo audio signal including the left and right channels; or to classify irrelevant stereo content in the stereo audio signal including the left and right channels in response to features extracted from the stereo audio signal including the left and right channels.

The present disclosure relates to a stereo sound decoding method and a stereo sound decoding system. The stereo sound decoding method includes receiving encoded parameters including encoded parameters of a primary channel and encoded parameters of a secondary channel, wherein the primary channel encoded parameters include LP filter coefficients of the primary channel; decoding the primary channel in response to the primary channel encoded parameters; and decoding the secondary channel using one of a plurality of encoding models, wherein (a) at least one of the encoding models uses the primary channel LP filter coefficients to decode the secondary channel, and (b) at least one of the encoding models uses primary channel encoded parameters other than the LP filter coefficients to decode the secondary channel.

A method and system for improving robustness of wireless communications is provided. The provided method and system transmit a data packet on a first isochronous stream and transmit one or more supplemental data packets in the same time interval. The one or more supplemental data packets are used to recreate and / or extend at least a portion of one or more data packets of a plurality of data packets already transmitted. Alternatively, the one or more supplemental data packets are used to create and / or extend at least a portion of one or more data packets of a plurality of data packets received during a next isochronous interval. The method and system described herein enables improved robustness by enabling better retransmissions using correctly received packets, and the method described herein works with any Bluetooth broadcaster sink without modification.

To provide a method and system for dynamic voice enhancement of an audio source.SOLUTION: A method of processing audio source input signals includes: first path signal processing including receiving an audio source input and performing dynamic loudness balancing on the audio source input based on a first gain control parameter; and second path signal processing including performing voice detection on the audio source input and calculating a detection confidence, and calculating a second gain control parameter based on the detection confidence. The first gain control parameter is updated with the second gain control parameter calculated by a second processing path. The first path signal processing and the second path signal processing may be performed synchronously or asynchronously.SELECTED DRAWING: Figure 8

The exemplary method comprises the step of receiving an audio signal associated with a virtual auditory space location. A digital filter is selected based on the virtual auditory space location. The digital filter includes one or more notch filters containing one or more center frequencies, based on a roughly sigmoid distribution of center frequencies as a function of the virtual auditory space location. The notch filters are configured to produce one or more notches in the frequency spectrum of the audio signal when applied to the audio signal. The digital filters are applied to the audio signal to obtain a processed audio signal. An output audio signal is generated based on the processed audio signal. The output audio signal is provided to the device to produce a virtual auditory display sound.

Disclosed is an apparatus for outputting audio data including a binaural bit, the apparatus including an information collector configured to obtain user information and a processor configured to control an output of audio data including a binaural bit, wherein the processor may be further configured to determine a volume level of an audio output of the binaural bit according to the user information obtained at a first timing point and output the audio data including the binaural bit according to the volume level determined at the first timing point and wherein the processor may be further configured to determine the volume level of the audio output of the binaural bit according to the user information obtained while the audio data is being outputted at a second timing point and output the audio data including the binaural bit according to the volume level determined at the second timing point.

A method (1900) for audio signal filtering. The method includes generating (s1902) a pair of filters for a certain location specified by an elevation angle ϑ and an azimuth angle φ, the pair of filters consisting of a right filter (ĥ r (ϑ, φ)) and a left filter (ĥ l (ϑ, φ)); filtering (s1904) an audio signal using the right filter; and filtering (s1906) the audio signal using the left filter. Generating the pair of filters comprises: i) obtaining at least a first set of elevation basis function values at the elevation angle; ii) obtaining at least a first set of azimuth basis function values at the azimuth angle; iii) generating the right filter using: a) at least the first set of elevation basis function values, b) at least the first set of azimuth basis function values, and c) right filter model parameters; and iv) generating the left filter using: a) at least the first set of elevation basis function values, b) at least the first set of azimuth basis function values, and c) left filter model parameters.

The present application provides a sound reproduction method, a program product, and a sound reproduction apparatus. The sound reproduction method includes: a signal acquisition step of acquiring a plurality of first audio signals and classification information, the first audio signals representing first sounds reaching a listener from a first range as a prescribed angular range, and the classification information being information in which the plurality of first audio signals are classified; and a correction processing step of, in a case where one of the plurality of first sounds is determined to be a trigger sound, implementing correction processing on a first audio signal representing the determined trigger sound and one or more first audio signals representing one or more first sounds classified into the same group as the trigger sound, among the plurality of acquired first audio signals, based on the acquired classification information.

An exemplary process for wirelessly streaming audio data to a head-mounted display (HMD) with low latency and reduced auditory distortion is described. This includes determining one or more statistics indicating the latency associated with audio data wirelessly received by the HMD and stored in a buffer within the HMD, and determining, at least in part, to adjust the size of the buffer based on these statistics. To reduce the buffer size, segments of audio data can be replaced with synthesized audio data segments to obtain modified audio data in the buffer. To increase the buffer size, synthesized audio data segments can be added to the audio data to obtain modified audio data in the buffer. Audio content can then be output via one or more speakers of the HMD, at least in part, based on the modified audio data.

According to an embodiment of the present invention, a device configured to output audio data including binaural beats is disclosed, and the device may comprise: an information collector configured to acquire user information; and a processor configured to control an output of audio data including binaural beats, wherein the processor is configured to: determine a volume level of an audio output of the binaural beats according to user information acquired at a first time point; output the audio data including the binaural beats according to the volume level determined at the first time point; determine a volume level of an audio output of the binaural beats according to user information acquired while the audio data is being output at a second time point; and output the audio data including the binaural beats according to the volume level determined at the second time point.

A method (200) of processing audio in an immersive audio format comprising at least one height audio channel, comprising: obtaining (250) two height audio signals from at least a portion of the at least one height audio channel: modifying (270) a relative phase between the two height audio signals in frequency bands in which phase differences are predominantly out of phase to obtain two phase modified height audio signals: and playing back (290) the processed audio comprising the two phase modified height audio signals with at least two audio loudspeakers. The phase differences occur as a result of having monaural signals emanated from two audio loudspeakers at one or more listening positions symmetrically off-center with respect to the at least two loudspeakers, laterally spaced with respect to each of said one or more listening positions. The method allows perception of sound height / elevation without using overhead speakers.

The present invention relates to an audio signal processing method, a parameterization device and an audio signal processing device for the same, and more particularly, to an audio signal processing method to implement filtering of an input audio signal with a low computational complexity, and a parameterization device and an audio signal processing device for the same. To this end, provided are a method for processing an audio signal, including: receiving an input audio signal; receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of the input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; truncating each subband filter coefficients based on filter order information obtained by at least partially using characteristic information extracted from the corresponding subband filter coefficients, the truncated subband filter coefficients being filter coefficients of which energy compensation is performed based on the flag information and the length of at least one truncated subband filter coefficients being different from the length of the truncated subband filter coefficients of another subband; and filtering each subband signal of the input audio signal by using the truncated subband filter coefficients, and a parameterization device and an audio signal processing device for the same.

This application provides a stereo signal encoding method and apparatus. The encoding method includes: determining a residual signal encoding parameter of a current frame of a stereo signal based on downmixed signal energy and residual signal energy of each of M sub-bands of the current frame, where the residual signal encoding parameter of the current frame is used to indicate whether to encode residual signals of the M sub-bands, the M sub-bands are at least some of N sub-bands, N is a positive integer greater than 1, M ≤ N, and M is a positive integer (301); and determining, based on the residual signal encoding parameter of the current frame, whether to encode the residual signals of the M sub-bands of the current frame (302). According to the encoding method, high-frequency distortion of a decoded stereo signal can be reduced as much as possible while a spatial sense and sound-image stability of the decoded stereo signal are improved, thereby improving encoding quality.

An audio system includes a first left speaker arranged at a position corresponding to a left side of a head of a user on a first seat, a first right speaker arranged at a position corresponding to a right side of the head of the user on the first seat, a second left speaker arranged at a position corresponding to a left side of a head of a user on a second seat disposed side by side with the first seat, a second right speaker arranged at a position corresponding to a right side of the head of the user on the second seat, a front speaker arranged at a central position between the first seat and the second seat in a lateral direction, the central position being in front of the first seat and the second seat, and acoustic signal processing circuitry.

The device includes receivers 101 and 102 that receive microphone and video signals capturing a scene. An image generator 113 generates an image signal containing an image generated from the video signal, and a zoom determiner 115 determines zoom characteristics such as the zoom coefficient of the image. An audio component generator 107 generates a stereo signal component of a given monaural audio signal, containing a monaural audio signal positioned at a certain location within the stereo image of the stereo signal component. A coupler 109 generates a stereo audio signal containing at least one stereo signal component, and an output unit 111 generates an audio-video signal containing the stereo audio signal and the image signal. In this approach, the position is determined based on zoom characteristics and typically also based on the direction of arrival of the given monaural audio signal. This approach may improve the consistency between visual and auditory spatial cues.

An equipment for monitoring physiological status of a user includes a sound detection unit, a chest detection unit, an abdomen detection unit, and a control unit. The sound detection unit is configured to detect sound of breath and to generate a sound detection signal based on the sound of breath. The chest detection unit is configured to detect a parameter related to movement of a chest and to generate a chest detection signal based on the detection. The abdomen detection unit is configured to detect a parameter related to movement of an abdomen and to generate an abdomen detection signal based on the detection. The control unit is electrically connected to the sound detection unit, the chest detection unit, and the abdomen detection unit and is configured to determine a breathing status of a user based on the sound detection signal, the chest detection signal, and the abdomen detection signal.

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 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.

An audio apparatus comprises a first renderer (205) applying a directional rendering to a mono downmix with the rendering being based on a channel transfer function retrieved from a store (211) storing transfer functions for different directions. The rendering is for a direction determined in response to spatial upmix parameters provided for the downmix by a direction determining circuit (209). A second intermediate stereo signal may be rendered from a decorrelated version of the mono downmix by a second renderer (217) using a predetermined rendering, such as for example from a specific direction / position that is not signal dependent. A combiner (219) generates an output stereo signal by combining the intermediate stereo signals. An adapter (221) adapts the level of the first intermediate stereo signal relative to the level of the second intermediate stereo signal in the output stereo signal dependent on a first scaling value determined from the spatial parameters.

An audio apparatus comprises a downmixer (107) generating a mono downmix by downmixing a stereo signal resulting from applying downmix filters (103, 105) to an input stereo signal. Residual filters (109,113) generate compensation signals from the mono downmix and residual signals are generated by compensators (111, 115) compensating the input channel signals by the compensation signals. A renderer 117 renders the mono downmix and residual signals to generate an output stereo signal. The filter responses of the downmix filter (103, 105) and the residual filters (109,113) for a given channel have a fixed relationship, typically being complex conjugates in the frequency domain, and are updated by an adapter (119) to reduce error values determined from the residual signals. The error values may be energy or power level measures for the residual signals.

An integrated circuit (IC) 100 includes a first transducer driver 102 and a second transducer driver 104, for respectively driving first and second transducers 114, 116, and boost converter circuitry

An audio personalisation method for simulating perception of a vertical displacement of a sound source, the method comprising the steps of: obtaining an input head related transfer function, HRTF, associated with a user; determining an intended vertical displacement for the sound source; selecting at least one frequency region in the input HRTF; and adjusting the amplitude of the selected frequency region(s) to simulate the intended vertical displacement for the sound source. This provides improvements to the generation and / or manipulation of HRTFs to allow adjustment of the perceived location of a sound source.

A multi-channel audio decoder for providing at least two output audio signals on the basis of an encoded representation is configured to perform a weighted combination of a downmix signal, a decorrelated signal and a residual signal, to obtain one of the output audio signals. The multi-channel audio decoder is configured to determine a weight describing a contribution of the decorrelated signal in the weighted combination in dependence on the residual signal. A multi-channel audio encoder for providing an encoded representation of a multi-channel audio signal is configured to obtain a downmix signal on the basis of the multi-channel audio signal, to provide parameters describing dependencies between the channels of the multi-channel audio signal, and to provide a residual signal. The multi-channel audio encoder is configured to vary an amount of residual signal included into the encoded representation in dependence on the multi-channel audio signal.

An audio apparatus comprises a first renderer (205) applying a directional rendering to a mono downmix with the rendering being based on a channel transfer function retrieved from a store (211) storing transfer functions for different directions. The rendering is for a direction determined in response to spatial upmix parameters provided for the downmix by a direction determining circuit (209). A second intermediate stereo signal may be rendered from a decorrelated version of the mono downmix by a second renderer (217) using a predetermined rendering, such as for example from a specific direction / position that is not signal dependent. A combiner (219) generates an output stereo signal by combining the intermediate stereo signals. An adapter (221) adapts the level of the first intermediate stereo signal relative to the level of the second intermediate stereo signal in the output stereo signal dependent on a first scaling value determined from the spatial parameters.

A data processing method includes acquiring video frame data including one or more video frames and audio data of a video, and determining position attribute information of a target object in the acquired one or more video frames, the target object being associated with the audio data. The method also includes acquiring a channel encoding parameter associated with the position attribute information, and performing azimuth enhancement processing on the audio data according to the channel encoding parameter to obtain enhanced audio data. Apparatus and non-transitory computer-readable storage medium counterpart embodiments are also contemplated.

A method for encoding an input audio stream including the steps of obtaining a first playback stream presentation of the input audio stream intended for reproduction on a first audio reproduction system, obtaining a second playback stream presentation of the input audio stream intended for reproduction on a second audio reproduction system, determining a set of transform parameters suitable for transforming an intermediate playback stream presentation to an approximation of the second playback stream presentation, wherein the transform parameters are determined by minimization of a measure of a difference between the approximation of the second playback stream presentation and the second playback stream presentation, and encoding the first playback stream presentation and the set of transform parameters for transmission to a decoder.

The application discloses a kind of audio channel independent control method and system under folding screen terminal split screen scene.The method is detected when folding screen terminal is in unfolded state and triggers split screen display operation, obtains split screen mode and the spatial position information of each split screen area, identifies the application instance respectively running in each split screen area, and establishes the association between application instance and corresponding split screen area position.Based on preset channel binding rule, the application instance corresponding to different split screen area is respectively bound to different audio output channel, the audio signal generated by each application instance is processed by instance level isolation, and the isolated audio signal is directed to the corresponding audio output channel;When split screen state or application instance changes, dynamically update channel binding relationship.The application can effectively avoid multi-application audio mixed output in split screen use scene, improve audio recognition and multi-task operation experience, and is suitable for the equipment with split screen display capability.