Abstract: An electro-acoustic transducer, comprising a supporting frame, a magnet assembly with an annular yoke surrounding a magnet a diaphragm attached to the front edge of the supporting frame, a voice coil suspended by the diaphragm in a gap formed between the magnet and the annular yoke, the voice coil being axially movable with respect to the magnet, and an annular damper arranged to stabilize the diaphragm. The transducer further comprises a damper holder having a substantially flat annular portion attached to the diaphragm, and a conical wall portion surrounding the voice coil, wherein an inner perimeter of the damper is attached to a rear region of the conical wall portion.

Abstract: The present invention is directed to methods and apparatus for translating a first plurality of audio input channels to a second plurality of audio output channels. This includes determining that there is pair-wise coding among any of the first plurality of audio input channels, determining an input/output-mapping matrix for mapping at least a first set of the first plurality of audio input channels to at least a second set of the second plurality of audio output channels; and deriving the second plurality of audio output channels based on first plurality of audio input channels, the input/output-mapping matrix and the determined pair-wise coding. The first plurality of audio input channels represent the same soundfield represented by the second plurality of audio output channels.

Abstract: The invention improves HOA sound field representation compression and decompression. A decoder decodes compressed dominant directional signals and compressed residual component signals so as to provide decompressed dominant directional signals and decompressed time domain signals representing a residual HOA component in a spatial domain. A re-correlator re-correlates the decompressed time domain signals to obtain a corresponding reduced-order residual HOA component. A processor determines a decompressed residual HOA component based on the corresponding reduced-order residual HOA component, and determines predicted directional signals based on at least a parameter. The processor is further configured to determine an HOA sound field representation based on the decompressed dominant directional signals, the predicted directional signals, and the decompressed residual HOA component.

Abstract: A technique including receiving and decoding a coded bitstream encoded with audio content including first audio objects corresponding to a first media content type of two consecutive media content types and second audio objects corresponding to a second media content type of the two consecutive media content types, and audio metadata corresponding to the audio content. The audio metadata including first and second audio object gains, for the first and second audio objects, generated in part based on a first fading curve of the first media content type and a second fading curve of the second media content type, respectively. The technique further includes applying the first and second audio object gains to the first and second audio objects, and rendering a sound field represented by the first audio object with the applied first audio object gain and the second audio object with the applied second audio object gain.

Abstract: An apparatus and method of interfacing between a source media device and a destination media device. A wireless module passes through an audio signal from the source device to an output device, and transmits a wireless signal to a wireless device that outputs the audio signal. In this manner, the number of devices used for the connections may be reduced.

Abstract: A method for generating a bitstream indicative of an object based audio program is described. The bitstream comprises a sequence of containers. A first container of the sequence of containers comprises a plurality of substream entities for a plurality of substreams of the object based audio program and a presentation section. The method comprises determining a set of object channels. The method further comprises providing a set of object related metadata for the set of object channels. In addition, the method comprises inserting a first set of object channel frames and a first set of object related metadata frames into a respective set of substream entities of the first container. Furthermore, the method comprises inserting presentation data into the presentation section.

Abstract: Improved methods and/or apparatus for decoding an encoded audio signal in soundfield format for L loudspeakers. The method and/or apparatus can render an Ambisonics format audio signal to 2D loudspeaker setup(s) based on a rendering matrix. The rendering matrix has elements based on loudspeaker positions and wherein the rendering matrix is determined based on weighting at least an element of a first matrix with a weighting factor g = 1 L . The first matrix is determined based on positions of the L loudspeakers and at least a virtual position of at least a virtual loudspeaker that is added to the positions of the L loudspeakers.

Abstract: The present disclosure relates to the field of audio coding, in particular, it relates to a method for encoding audio signals through a masking model based on a hearing threshold of frequency intervals of the audio signal and a measured energy of the audio signal for the corresponding frequency intervals. The disclosure further relates to an encoder that is capable of carrying out the audio encoding method.

Abstract: A handheld imaging device has a data receiver that is configured to receive reference encoded image data. The data includes reference code values, which are encoded by an external coding system. The reference code values represent reference gray levels, which are being selected using a reference grayscale display function that is based on perceptual non-linearity of human vision adapted at different light levels to spatial frequencies. The imaging device also has a data converter that is configured to access a code mapping between the reference code values and device-specific code values of the imaging device. The device-specific code values are configured to produce gray levels that are specific to the imaging device. Based on the code mapping, the data converter is configured to transcode the reference encoded image data into device-specific image data, which is encoded with the device-specific code values.

Abstract: The present document describes a method (400) for personalizing audio content. The method (400) comprises receiving (401) a manifest file (140) for the audio content. The manifest file (140) comprises at least one adaptation set (281, 282) referencing an audio bitstream (121), where the audio bitstream (121) comprises a plurality of audio objects (181), and a plurality of different preselection elements (291, 292, 293) for the adaptation set (281, 282), wherein the different preselection elements (291, 292, 293) specify different combinations of the plurality of audio objects (181). The method (400) further comprises selecting (402) a preselection element (291) from the plurality of different preselection elements (291, 292, 293), and causing (403) rendering of an audio signal which depends on the selected preselection element (291).

Abstract: A digital PSF for use in a dual modulation display. The invention allows the use of less than optimal point spread (PSF) functions in the optics between the pre-modulator and primary modulator of a dual modulation projection system. This technique uses multiple halftones per frame in the pre-modulator synchronized with a modified bit sequence in the primary modulator to produce a compensation image that reduces the errors produced by the sub-optimal PSF. The invention includes the application to dual modulation and dual modulated 3D viewing systems.

Abstract: There are two representations for Higher Order Ambisonics denoted HOA: spatial domain and coefficient domain. The invention generates from a coefficient domain representation a mixed spatial/coefficient domain representation, wherein the number of said HOA signals can be variable. An aspect of the invention further relates to methods and apparatus decoding multiplexed and perceptually encoded HOA signals, including transforming a vector of PCM encoded spatial domain signals of the HOA representation to a corresponding vector of coefficient domain signals by multiplying the vector of PCM encoded spatial domain signals with a transform matrix and de-normalizing the vector of PCM encoded and normalized coefficient domain signals, wherein said de-normalizing comprises.

Abstract: Described is a method of hosting a teleconference among a plurality of client devices arranged in two or more acoustic spaces, each client device having an audio capturing capability and/or an audio rendering capability, the method comprising: grouping the plurality of client devices into two or more groups based on their belonging to respective acoustic spaces, receiving first audio streams from the plurality of client devices, generating second audio streams from the first audio streams for rendering by respective client devices among the plurality of client devices, based on the grouping of the plurality of client devices into the two or more groups, and outputting the generated second audio streams to respective client devices. Further described are corresponding computation devise, computer programs, and computer-readable storage media.

Abstract: A method for mitigating modal noise includes applying a time-varying mechanical force to a fiber segment of the multimode optical fiber in at least a first direction orthogonal to a fiber axis of the multimode optical fiber within the fiber segment. A modal-noise mitigator for a multimode optical fiber includes an actuator configured to apply a time-varying mechanical force to a fiber segment of the multimode optical fiber in at least a first direction orthogonal to a fiber axis of the multimode optical fiber within the fiber segment.

Abstract: Overlapped block disparity estimation and compensation is described. Compensating for images with overlapped block disparity compensation (OBDC) involves determining if OBDC is enabled in a video bit stream, and determining if OBDC is enabled for one or more macroblocks that neighbor a first macroblock within the video bit stream. The neighboring macroblocks may be transform coded. If OBDC is enabled in the video bit stream and for the one or more neighboring macroblocks, predictions may be made for a region of the first macroblock that has an edge adjacent with the neighboring macroblocks. OBDC can be causally applied. Disparity compensation parameters or modes may be shared amongst views or layers. A variety of predictions may be used with causally-applied OBDC.

Abstract: Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

Abstract: Disclosed is a data transmission system that transmits data by using a relay. The relay selects a transmission terminal from among a plurality of terminals accessing a base station. A base station transmits base station data to the relay during a first time slot, and the transmission terminal transmits terminal data to the relay. The relay transmits terminal data to the base station during a second time slot, and transmits base station data to the transmission terminal.

Abstract: Computer-implemented methods for speech synthesis are provided. A speech synthesizer may be trained to generate synthesized audio data that corresponds to words uttered by a source speaker according to speech characteristics of a target speaker. The speech synthesizer may be trained by time-stamped phoneme sequences, pitch contour data and speaker identification data. The speech synthesizer may include a voice modeling neural network and a conditioning neural network.

Abstract: The present disclosure relates to a method of decoding audio scene content from a bitstream by a decoder that includes an audio renderer with one or more rendering tools.

Abstract: Improved methods and/or apparatus for decoding an encoded audio signal in soundfield format for L loudspeakers. The method and/or apparatus can render an Ambisonics format audio signal to 2D loudspeaker setup(s) based on a rendering matrix. The rendering matrix has elements based on loudspeaker positions and wherein the rendering matrix is determined based on weighting at least an element of a first matrix with a weighting factor g = 1 L . The first matrix is determined based on positions of the L loudspeakers and at least a virtual position of at least a virtual loudspeaker that is added to the positions of the L loudspeakers.

Abstract: A speech separation server comprises a deep-learning encoder with nonlinear activation. The encoder is programmed to take a mixture audio waveform in the time domain, learn generalized patterns from the mixture audio waveform, and generate an encoded representation that effectively characterizes the mixture audio waveform for speech separation.

Abstract: A set of tensor-product B-Spline (TPB) basis functions is determined. A set of selected TPB prediction parameters to be used with the set of TPB basis functions for generating predicted image data in mapped images from source image data in source images of a source color grade is generated. The set of selected TPB prediction parameters is generated by minimizing differences between the predicted image data in the mapped images and reference image data in reference images of a reference color grade. The reference images correspond to the source images and depict same visual content as depicted by the source images. The set of selected TPB prediction parameters is encoded in a video signal as a part of image metadata along with the source image data in the source images. The mapped images are caused to be reconstructed and rendered with a recipient device of the video signal.

Abstract: Embodiments are disclosed for bitrate distribution in immersive voice and audio services. In an embodiment, a method of encoding an IVAS bitstream comprises: receiving an input audio signal; downmixing the input audio signal into one or more downmix channels and spatial metadata; reading a set of one or more bitrates for the downmix channels and a set of quantization levels for the spatial metadata from a bitrate distribution control table; determining a combination of the one or more bitrates for the downmix channels; determining a metadata quantization level from the set of metadata quantization levels using a bitrate distribution process; quantizing and coding the spatial metadata using the metadata quantization level; generating, using the combination of one or more bitrates, a downmix bitstream for the one or more downmix channels; combining the downmix bitstream, the quantized and coded spatial metadata and the set of quantization levels into the IVAS bitstream.

Abstract: Some implementations involve receiving a content stream that includes audio data, determining a content type corresponding to the content stream and determining, based at least in part on the Receiving, by a control system and via an interface system, a content stream that includes audio data content type, a noise compensation method. Some examples involve performing the noise compensation method on the audio data to produce noise-compensated audio data, rendering the noise-compensated audio data for reproduction via a set of audio reproduction transducers of the audio environment, to produce rendered audio signals, and providing the rendered audio signals to at least some audio reproduction transducers of the audio environment.

Abstract: Audio content coded for a reference speaker configuration is downmixed to downmix audio content coded for a specific speaker configuration. One or more gain adjustments are performed on individual portions of the downmix audio content coded for the specific speaker configuration. Loudness measurements are then performed on the individual portions of the downmix audio content. An audio signal that comprises the audio content coded for the reference speaker configuration and downmix loudness metadata is generated. The downmix loudness metadata is created based at least in part on the loudness measurements on the individual portions of the downmix audio content.

Abstract: Methods, systems, and bitstream syntax are described for canvas size, single layer or multi-layer, scalable decoding, with support for regions of interest (ROI), using a decoder supporting reference picture resampling. Offset parameters for a region of interest in a current picture and offset parameters for an ROI in a reference picture are taken into consideration when computing scaling factors to apply reference picture resampling. Syntax elements for supporting ROI regions under reference picture resampling are also presented.

Abstract: The present disclosure relates to an apparatus for decoding an encoded Unified Audio and Speech stream. The apparatus comprises a core decoder for decoding the encoded Unified Audio and Speech stream. The core decoder includes a fast Fourier transform, FFT, module implementation based on a Cooley-Tuckey algorithm. The FFT module is configured to determine a discrete Fourier transform, DFT. Determining the DFT involves recursively breaking down the DFT into small FFTs based on the Cooley-Tucker algorithm and using radix-4 if a number of points of the FFT is a power of 4 and using mixed radix if the number is not a power of 4. Performing the small FFTs involves applying twiddle factors. Applying the twiddle factors involves referring to pre-computed values for the twiddle factors.

Abstract: Methods and systems for generating an image quality metric are described. A reference and a test image are first converted to the ITP color space. After calculating difference images ?I, ?T, and ?P, using the color channels of the two images, the difference images are convolved with low pass filters, one for the I channel and one for the chroma channels (I or P). The image quality metric is computed as a function of the sum of squares of filtered ?I, ?T, and ?P values. The chroma low-pass filter is designed to maximize matching the image quality metric with subjective results.

Abstract: An acoustic transducer that includes a housing, a diaphragm, a spider, a motor, and a drop ring. The motor includes a backplate, a frontplate, a magnet, and a voice coil. The drop ring connects the diaphragm to the spider at a circumference of the spider. The drop ring extends parallel with respect to a central axis of the housing. The circumference of the spider is spaced away from the motor and connects to the diaphragm at a resonant node of the diaphragm.

Abstract: A method for representing a second presentation of audio channels or objects as a data stream, the method comprising the steps of: (a) providing a set of base signals, the base signals representing a first presentation of the audio channels or objects; (b) providing a set of transformation parameters, the transformation parameters intended to transform the first presentation into the second presentation; the transformation parameters further being specified for at least two frequency bands and including a set of multi-tap convolution matrix parameters for at least one of the frequency bands.

Abstract: Embodiments for a speaker system that produces a near-field sound pattern for rendering immersive audio content in a portable device. An array of drivers projects sound upwards from a top surface of the portable device to form upward-firing speakers; a set of speakers projects sound downwards from a bottom surface of the portable device to form downward-firing speakers. A decoder/renderer component receives immersive audio content, decodes height audio signals from the content and sends direct audio signals to the downward-firing speakers. A crossover performs a high-pass filter function to pass high frequency components of the decoded height audio signals to the upward-firing speakers and low frequency components of the decoded height audio signals to the downward-firing speakers.

Abstract: A method for decoding an encoded audio bitstream is disclosed. The method includes receiving the encoded audio bitstream and decoding the audio data to generate a decoded lowband audio signal. The method further includes extracting high frequency reconstruction metadata and filtering the decoded lowband audio signal with an analysis filterbank to generate a filtered lowband audio signal. The method also includes extracting a flag indicating whether either spectral translation or harmonic transposition is to be performed on the audio data and regenerating a highband portion of the audio signal using the filtered lowband audio signal and the high frequency reconstruction metadata in accordance with the flag. The high frequency regeneration is performed as a post-processing operation with a delay of 3010 samples per audio channel.

Abstract: Described herein is a method of waveform decoding, the method including the steps of: (a) receiving, by a waveform decoder, a bitstream including a finite bitrate representation of a source signal; (b) waveform decoding the finite bitrate representation of the source signal to obtain a waveform approximation of the source signal; (c) providing the waveform approximation of the source signal to a generative model that implements a probability density function, to obtain a probability distribution for a reconstructed signal of the source signal; and (d) generating the reconstructed signal of the source signal based on the probability distribution. Described are further a method and system for waveform coding and a method of training a generative model.

Abstract: A height channel speaker with an integrated acoustic reflector to reflect sound off of a ceiling down to a listener. The acoustic reflector compensates for thin transducers by creating a virtual image of the real sound source outside the speaker enclosure. The focal point of the acoustic reflector is controlled by modifying the curvature of the reflector surface. The transducer is mounted on an inclined plane to radiate sound in a rear-upward inclined direction. The acoustic reflector is mounted on the same inclined plane so that the radiant axis of the transducer is directly incident on the acoustic reflector surface. The sound is projected towards the ceiling in a forward, upward-inclined direction to reflect off the ceiling and down to the listener. The speaker can be acoustically occluded from the listener by a panel to which the speaker is attached.

Abstract: The disclosure relates to methods of processing a spatial audio signal for generating a compressed representation of the spatial audio signal. The methods include analyzing the spatial audio signal to determine directions of arrival for one or more audio elements; for at least one frequency subband, determining respective indications of signal power associated with the directions of arrival; generating metadata including direction information that includes indications of the directions of arrival of the audio elements, and energy information that includes respective indications of signal power; generating a channel-based audio signal with a predefined number of channels based on the spatial audio signal; and outputting, as the compressed representation, the channel-based audio signal and the metadata.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: Methods and systems for frame rate scalability are described. Support is provided for input and output video sequences with variable frame rate and variable shutter angle across scenes, or for input video sequences with fixed input frame rate and input shutter angle, but allowing a decoder to generate a video output at a different output frame rate and shutter angle than the corresponding input values. Techniques allowing a decoder to decode more computationally-efficiently a specific backward compatible target frame rate and shutter angle among those allowed are also presented.

Abstract: Some implementations may involve receiving, via an interface system, personnel location data indicating a location of at least one person and receiving, from an orientation system, headset orientation data corresponding with the orientation of a headset. First environmental element location data, indicating a location of at least a first environmental element, may be determined. Based at least in part on the headset orientation data, the personnel location data and the first environmental element location data, headset coordinate locations of at least one person and at least the first environmental element in a headset coordinate system corresponding with the orientation of the headset may be determined. An apparatus may be caused to provide spatialization indications of the headset coordinate locations. Providing the spatialization indications may involve controlling a speaker system to provide environmental element sonification corresponding with at least the first environmental element location data.

Abstract: Methods and systems for canvas size scalability across the same or different bitstream layers of a video coded bitstream are described. Offset parameters for a conformance window, a reference region of interest (ROI) in a reference layer, and a current ROI in a current layer are received. The width and height of a current ROI and a reference ROI are computed based on the offset parameters and they are used to generate a width and height scaling factor to be used by a reference picture resampling unit to generate an output picture based on the current ROI and the reference ROI.

Abstract: A method is disclosed herein for adapting parameters of a sibilance detector. Time-frequency features are extracted from an audio signal being received and. Based on those time-frequency features, a determination is made of whether the audio signal includes a short-term feature or a long-term feature. In accordance with determining that the audio signal includes the short-term feature or the long-term feature, one or more parameters of a sibilance detector for detecting sibilance in the audio signal are adapted. Sibilance in the audio signal, is detected using the sibilance detector with the one or more adapted parameters.

Abstract: An audio object including audio content and object metadata is received. The object metadata indicates an object spatial position of the audio object to be rendered by audio speakers in a playback environment. Based on the object spatial position and source spatial positions of the audio speakers, initial gain values for the audio speakers are determined. The initial gain values can be used to select a set of audio speakers from among the audio speakers. Based on the object spatial position and a set of source spatial positions at which the set of audio speakers are respectively located in the playback environment, a set of non-negative optimized gain values for the set of audio speakers is determined. The audio object at the object spatial position is rendered with the set of optimized gain values for the set of audio speakers.

Abstract: The method for decoding an intra-picture prediction mode includes the steps of: determining whether the intra-picture prediction mode of a current prediction unit is identical to a first intra-picture prediction mode candidate or a second intra-picture prediction mode candidate based on bit information; and when the intra-picture prediction mode of the current prediction unit is identical to the first intra-picture prediction mode candidate and/or to the second intra-picture prediction mode candidate, determining whether the first intra-picture prediction mode candidate or the second intra-picture prediction mode candidate is identical to the intra-picture prediction mode of the current prediction unit on the basis of additional bit information, and decoding the intra-picture prediction mode of the current prediction unit.

Abstract: The present disclosure relates to a method of processing audio content including directivity information for at least one sound source, the directivity information comprising a first set of first directivity unit vectors representing directivity directions and associated first directivity gains. The disclosure further relates to corresponding methods of encoding and decoding audio content including directivity information for at least one sound source.

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component ({tilde over (C)}AMB(k?1)). The ambient HOA component ({tilde over (C)}AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.

Abstract: The disclosed embodiments enable converting audio signals captured in various formats by various capture devices into a limited number of formats that can be processed by an audio codec (e.g., an Immersive Voice and Audio Services (IVAS) codec). In an embodiment, a simplification unit of the audio device receives an audio signal captured by one or more audio capture devices coupled to the audio device. The simplification unit determines whether the audio signal is in a format that is supported/not supported by an encoding unit of the audio device. Based on the determining, the simplification unit, converts the audio signal into a format that is supported by the encoding unit. In an embodiment, if the simplification unit determines that the audio signal is in a spatial format, the simplification unit can convert the audio signal into a spatial “mezzanine” format supported by the encoding.

Abstract: An embodiment of the disclosure provides a method and a system to sense a light source based on a viewer position in relation to display device. The system receives sensor data from one or more light sensors mounted on a wearable device worn by a viewer of a display device in a room, where a field of view for the light sensors covers at least a field of view of the viewer. The system identifies a light source perceived in a field of view of the viewer based on the sensor data. The system transmits data for one or more operations to be performed by the display device displaying content to the viewer to compensate for a change in brightness or color of the content caused by the light source based at least in part on light source information of the light source.

Abstract: There are provided decoding and encoding methods for encoding and decoding of multichannel audio content for playback on a speaker configuration with N channels. The decoding method comprises decoding, in a first decoding module, M input audio signals into M mid signals which are suitable for playback on a speaker configuration with M channels; and for each of the N channels in excess of M channels, receiving an additional input audio signal corresponding to one of the M mid signals and decoding the input audio signal and its corresponding mid signal so as to generate a stereo signal including a first and a second audio signal which are suitable for playback on two of the N channels of the speaker configuration.

Abstract: A picture coding method of the present invention codes a picture signal and a ratio of a number of luminance pixels and a number of chrominance pixels for the picture signal, and then one coding method out of at least two coding methods is selected depending on the ratio. Next, data related to a picture size is coded in accordance with the selected coding method. The data related to the picture size indicates a size of the picture corresponding to the picture signal or an output area, which is a pixel area to be outputted in decoding in a whole pixel area coded in the picture signal coding.

Abstract: Disclosed are an apparatus and a method of encoding/decoding a video, particularly a method and an apparatus for storing a quantization parameter differential value in a largest coding unit (LCU) based on quadtree splitting and adaptively predicting a quantization parameter value based on context information on neighboring CUs. Quadtree-based quantization parameter encoding and decoding methods and apparatuses effectively show information on a block having a quantization parameter differential value based on splitting information on a CU and adaptively predict a quantization parameter value using context information including a block size, block partition and a quantization parameter of a neighboring CU.