Abstract: Improved methods and devices for processing low-frequency audio data are provided. A bass extraction process may involve applying low-pass filters to received audio object signals, to produce extracted low-frequency audio signals. The bass extraction process may be performed prior to a process of rendering audio objects into speaker feed signals. A bass management process may involve routing the extracted low-frequency audio signals to the one or more speakers capable of reproducing low-frequency audio signals.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener (4) comprises apparatus (10) to deliver the audio soundfields to the listener (4). The delivery apparatus (10) delivers the audio soundfields to the listener (4) with first and second audio elements perceived by the listener (4) as emanating from first and second virtual source locations (1), respectively, and with the first audio element and/or the second audio element delivered to the listener (4) from a third virtual source location (3). The first virtual source location (1) and the second virtual source location (2) are perceived by the listener (4) as being located to the front of the listener (4), and the third virtual source location (3) is located to the rear or the side of the listener (4).

Abstract: A method for performing linear mixing on coupled Head-related transfer functions (HRTFs) to determine an interpolated HRTF for any specified arrival direction in a range (e.g., a range spanning at least 60 degrees in a plane, or a full range of 360 degrees in a plane), where the coupled HRTFs have been predetermined to have properties such that linear mixing can be performed thereon (to generate interpolated HRTFs) without introducing significant comb filtering distortion. In some embodiments, the method includes steps of: in response to a signal indicative of a specified arrival direction, performing linear mixing on data indicative of coupled HRTFs of a coupled HRTF set to determine an HRTF for the specified arrival direction; and performing HRTF filtering on an audio input signal using the HRTF for the specified arrival direction.

Abstract: The present document relates to audio conference systems. In particular, the present document relates to the mapping of soundfields within an audio conference system. A conference multiplexer (110, 175, 210, 400) configured to place a first input soundfield signal (402) originating from a first soundfield endpoint (120, 170) within a 2D or 3D conference scene (300) to be rendered to a listener (301) is described. The first input soundfield signal (402) is indicative of a soundfield captured by the first soundfield endpoint (120, 170).

Abstract: A spread spectrum data hiding for audio signals is described. A set of pseudo-random noise sequences is added to an audio signal according to a data to be embedded. A masking curve is used to shape the added noise. A transient detection step can be used to control whether a shaped noise sequence is to be added or not. Embedded information is detected by first performing a whitening step and then performing a phase-only correlation with a same set of pseudo-random noise sequences. A detection method that is based on correlation of multiplexed noise sequences with a noise sequence embedded in the audio is also described.

Abstract: Embodiments of the present invention relate to audio signal processing. Specifically, a method for processing audio signal is provided, the method comprising: determining, for a current frame of the audio signal, frequency band energies for a plurality of predefined frequency bands at least partially based on frequency parameters of the current frame; generating frequency band gains for the plurality of predefined frequency bands by processing the frequency band energies; and generating frequency bin gains for the current frame based on the frequency band gains using predefined frequency band filter banks, the frequency band filter banks being specific to the plurality of predefined frequency bands. Corresponding system and computer program product are also disclosed.

Abstract: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: A spatial audio processing system and method including the steps of: dividing the series of virtual speakers into a series of horizontal planes around the expected listener; rendering the audio source to an intermediate spatial format for playback over a series of virtual speakers arranged in each of the series of planes around the listener, the rendering including: an initial panning of the spatialized virtual audio source to each of the horizontal planes to produce a plane rendered audio emission; a subsequent panning of each of the plane rendered audio emissions to a series of virtual speaker locations within each plane, with the subsequent panning utilising a series of panning curves which are spatially smoothed to can include spatial frequency components which are less than the Nyquist sampling rate of the audio source.

Abstract: Embodiments are directed to a method of representing spatial rendering metadata for processing in an object-based audio system that allows for lossless interpolation and/or re-sampling of the metadata. The method comprises time stamping the metadata to create metadata instances, and encoding an interpolation duration to with each metadata instance that specifies the time to reach a desired rendering state for the respective metadata instance. The re-sampling of metadata is useful for re-clocking metadata to an audio coder and for the editing audio content.

Abstract: Improved methods and devices for processing low-frequency audio data are provided. A bass extraction process may involve applying low-pass filters to received audio object signals, to produce extracted low-frequency audio signals. The bass extraction process may be performed prior to a process of rendering audio objects into speaker feed signals. A bass management process may involve routing the extracted low-frequency audio signals to the one or more speakers capable of reproducing low-frequency audio signals.

Abstract: An audio playback system generates output signals for multiple channels of acoustic transducers by applying a rendering matrix to data representing the aural content and spatial characteristics of audio objects, so that the resulting sound field creates accurate listener impressions of the spatial characteristics. Matrix coefficients are updated to render moving objects. Discontinuous updates of the rendering matrix coefficients are controlled according to psychoacoustic principles to reduce audible artifacts. The updates may also be managed to control the amount of data needed to perform the updates.

Abstract: An apparatus may include an inner module, an outer module that substantially surrounds a perimeter of the inner module, a plurality of light emitters, and a light distribution medium. The plurality of light emitters may be positioned under the inner module and project light radially outward. The light distribution medium may transport the light projected from the plurality of light emitters to an edge of the light distribution medium. The edge may include a diffusive surface and traverse a substantial portion of a boundary between the inner module and the outer module.

Abstract: A spread spectrum data hiding for audio signals is described. A set of pseudo-random noise sequences is added to an audio signal according to a data to be embedded. A masking curve is used to shape the added noise. A transient detection step can be used to control whether a shaped noise sequence is to be added or not. Embedded information is detected by first performing a whitening step and then performing a phase-only correlation with a same set of pseudo-random noise sequences. A detection method that is based on correlation of multiplexed noise sequences with a noise sequence embedded in the audio is also described.

Abstract: A spread spectrum data hiding for audio signals is described. A set of pseudo-random noise sequences is added to an audio signal according to a data to be embedded. A masking curve is used to shape the added noise. A transient detection step can be used to control whether a shaped noise sequence is to be added or not. Embedded information is detected by first performing a whitening step and then performing a phase-only correlation with a same set of pseudo-random noise sequences. A detection method that is based on correlation of multiplexed noise sequences with a noise sequence embedded in the audio is also described.

Abstract: A method for performing linear mixing on coupled Head-related transfer functions (HRTFs) to determine an interpolated HRTF for any specified arrival direction in a range (e.g., a range spanning at least 60 degrees in a plane, or a full range of 360 degrees in a plane), where the coupled HRTFs have been predetermined to have properties such that linear mixing can be performed thereon (to generate interpolated HRTFs) without introducing significant comb filtering distortion. In some embodiments, the method includes steps of: in response to a signal indicative of a specified arrival direction, performing linear mixing on data indicative of coupled HRTFs of a coupled HRTF set to determine an HRTF for the specified arrival direction; and performing HRTF filtering on an audio input signal using the HRTF for the specified arrival direction.

Abstract: A computer implemented system for rendering captured audio soundfields to a listener (4) comprises apparatus (10) to deliver the audio soundfields to the listener (4). The delivery apparatus (10) delivers the audio soundfields to the listener (4) with first and second audio elements perceived by the listener (4) as emanating from first and second virtual source locations (1), respectively, and with the first audio element and/or the second audio element delivered to the listener (4) from a third virtual source location (3). The first virtual source location (1) and the second virtual source location (2) are perceived by the listener (4) as being located to the front of the listener (4), and the third virtual source location (3) is located to the rear or the side of the listener (4).

Abstract: Sound source localization apparatuses and methods are described. A frame amplitude difference vector is calculated based on short time frame data acquired through an array of microphones. The frame amplitude difference vector reflects differences between amplitudes captured by microphones of the array during recording the short time frame data. Similarity between the frame amplitude difference vector and each of a plurality of reference frame amplitude difference vectors is evaluated. Each of the plurality of reference frame amplitude difference vectors reflects differences between amplitudes captured by microphones of the array during recording sound from one of a plurality of candidate locations. A desired location of sound source is estimated based at least on the candidate locations and associated similarity. The sound source localization can be performed based at least on amplitude difference.

Abstract: In some embodiments, a method for generating a matrix-encoded two-channel audio signal in response to a horizontal B-format signal by performing a mixing operation. In other embodiments, a method for generating a matrix-encoded two-channel audio signal, including steps of generating microphone output signals (by capturing sound with a microphone array), and performing a mixing operation on the microphone output signals, where the mixing operation is equivalent to generating a horizontal B-format signal in response to the microphone output signals, and generating the matrix-encoded two-channel audio signal in response to the horizontal B-format signal. The microphone array is typically a small array of cardiod microphones (e.g., an array consisting of three cardiod microphones). Other aspects include systems (e.g., encoders) programmed or otherwise configured to perform any embodiment of the method for generating a matrix-encoded two-channel audio signal.

Abstract: The present document relates to audio conference systems. In particular, the present document relates to the mapping of soundfields within an audio conference system. A conference multiplexer (110, 175, 210, 400) configured to place a first input soundfield signal (402) originating from a first soundfield endpoint (120, 170) within a 2D or 3D conference scene (300) to be rendered to a listener (301) is described. The first input soundfield signal (402) is indicative of a soundfield captured by the first soundfield endpoint (120, 170).