Patent classifications
H04S2400/15
Method, system and computer program product for recording and interpolation of ambisonic sound fields
A method of recording ambisonic sound fields with a spatially distributed plurality of ambisonic microphones comprising a step of recording sound signals from plurality of ambisonic microphones a step of converting recorded sound signals to ambisonic sound fields and a step of interpolation of the ambisonic sound fields according to the invention comprises a step of generating synchronizing signals for particular ambisonic microphones for synchronized recording of sound signals from plurality of ambisonic microphones and during the step of interpolation of the ambisonic sound fields it includes filtering sound signals from particular microphones with individual filters having a distance-dependent impulse response having a cut-off frequency f.sub.c(d.sub.m) depending on distance d.sub.m between point of interpolation and m-th microphone applying gradual distance dependent attenuation applying re-balancing with amplification of 0.sup.th ordered ambisonic component and attenuating remaining ambisonic components. Invention further concerns recording system and computer program product.
Systems and methods for classifying beamformed signals for binaural audio playback
The disclosed computer-implemented method may include receiving a signal for each channel of an audio transducer array on a wearable device. The method may also include calculating a beamformed signal for each beam direction of a set of beamforming filters for the wearable device. Additionally, the method may include classifying a first beamformed signal from the calculated beamformed signals into a first class of sound and a second beamformed signal from the calculated beamformed signals into a second class of sound. The method may also include adjusting, based on the classifying, a gain of the first beamformed signal relative to the second beamformed signal. Furthermore, the method may include converting the beamformed signals into spatialized binaural audio based on a position of a user. Finally, the method may include transmitting the spatialized binaural audio to a playback device. Various other methods, systems, and computer-readable media are also disclosed.
Transform ambisonic coefficients using an adaptive network
A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device also includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are also configured to apply one adaptive network, based on a constraint, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.
Determination of composite acoustic parameter value for presentation of audio content
Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.
MIXED REALITY SYSTEM FOR CONTEXT-AWARE VIRTUAL OBJECT RENDERING
A computer-implemented method in conjunction with mixed reality gear (e.g., a headset) includes imaging a real scene encompassing a user wearing a mixed reality output apparatus. The method includes determining data describing a real context of the real scene, based on the imaging; for example, identifying or classifying objects, lighting, sound or persons in the scene. The method includes selecting a set of content including content enabling rendering of at least one virtual object from a content library, based on the data describing a real context, using various selection algorithms. The method includes rendering the virtual object in the mixed reality session by the mixed reality output apparatus, optionally based on the data describing a real context (“context parameters”). An apparatus is configured to perform the method using hardware, firmware, and/or software.
SPATIAL AUDIO FOR INTERACTIVE AUDIO ENVIRONMENTS
Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. A first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the sound source in the virtual environment, and the first intermediate audio signal is associated with a first bus. A second intermediate audio signal is determined. The second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment. The second intermediate audio signal is determined based on a location of the sound source, and further based on an acoustic property of the virtual environment. The second intermediate audio signal is associated with a second bus. The output audio signal is presented to the listener via the first and second buses.
AI BASED REMIXING OF MUSIC: TIMBRE TRANSFORMATION AND MATCHING OF MIXED AUDIO DATA
The present invention provides a method for processing audio data, comprising the steps of providing input audio data containing a mixture of audio data including first audio data of a first musical timbre and second audio data of a second musical timbre different from said first musical timbre, decomposing the input audio data to provide decomposed data representative of the first audio data, transforming the decomposed data to obtain third audio data.
INFORMATION PROCESSING DEVICE, INFORMATION PROCESSING METHOD, INFORMATION PROCESSING PROGRAM, AND INFORMATION PROCESSING SYSTEM
The information processing device includes an acquisition unit, a determination unit, and a signal processing unit. The acquisition unit acquires one or a plurality of other sound elements. The determination unit determines importance levels of the sound elements acquired by the acquisition unit. The signal processing unit changes a sound source position of at least one of a sound element of a content being reproduced or another sound element according to the importance levels of the sound elements determined by the determination unit. Consequently, even in a case where the sound element of the reproduced content and the another sound element are simultaneously generated, sound interference between the sound element of the reproduced content and the another sound element can be suppressed.
DUAL LISTENER POSITIONS FOR MIXED REALITY
A method of presenting audio comprises: identifying a first ear listener position and a second ear listener position in a mixed reality environment; identifying a first virtual sound source in the mixed reality environment; identifying a first object in the mixed reality environment; determining a first audio signal in the mixed reality environment, wherein the first audio signal originates at the first virtual sound source and intersects the first ear listener position; determining a second audio signal in the mixed reality environment, wherein the second audio signal originates at the first virtual sound source, intersects the first object, and intersects the second ear listener position; determining a third audio signal based on the second audio signal and the first object; presenting, to a first ear of a user, the first audio signal; and presenting, to a second ear of the user, the third audio signal.
Method and Apparatus for Low Complexity Low Bitrate 6DOF HOA Rendering
An apparatus for generating an immersive audio scene, the apparatus including circuitry configured to: obtain audio scene based sources, the audio scene based sources are associated with one or more positions in an audio scene, wherein each audio scene based source includes at least one spatial parameter and at least one audio signal; determine at least one position associated with at least one of the audio scene based sources; generate at least one audio source based on the determined at least one position, wherein the circuitry is configured to: generate at least one spatial audio parameter; and generate at least one audio source signal; and generate information about a relationship between the generated at least one spatial audio parameter and the at least one audio signals and the generated at least one audio source is selected based on a renderer preference.