Abstract: Some implementations involve receiving, from a first subband domain acoustic echo canceller (AEC) of a first audio device in an audio environment, first adaptive filter management data from each of a plurality of first adaptive filter management modules, each first adaptive filter management module corresponding to a subband of the first subband domain AEC, each first adaptive filter management module being configured to control a first plurality of adaptive filters. The first plurality of adaptive filters may include at least a first adaptive filter type and a second adaptive filter type. Some implementations involve extracting, from the first adaptive filter management data, a first plurality of extracted features corresponding to a plurality of subbands of the first subband domain AEC and estimating a current local acoustic state based, at least in part, on the first plurality of extracted features.

Abstract: Methods and systems of improving bass response for a speaker in a portable computing device are described. One portable computing device includes first and second cover parts that are joined together to form a casing of the portable computing device, wherein a speaker volume is formed between portions of the first and second cover parts; a speaker arranged within the speaker volume; and one or more elastic spacers arranged between the first and second cover parts. The one or more elastic spacers are arranged to counteract, by their elastic recoil forces, a compression of the speaker volume when the first and second cover parts are under external compressing forces. The one or more elastic spacers are arranged between the first and second cover parts to be partially compressed by the first and second cover parts in the absence of external compressing forces on the first and second cover parts.

Abstract: An encoding device evaluates a plurality of processing and/or post-processing algorithms and/or methods to be applied to a video stream, and signals a selected method, algorithm, class or category of methods/algorithms either in an encoded bitstream or as side information related to the encoded bitstream. A decoding device or post-processor utilizes the signaled algorithm or selects an algorithm/method based on the signaled method or algorithm. The selection is based, for example, on availability of the algorithm/method at the decoder/post-processor and/or cost of implementation. The video stream may comprise, for example, downsampled multiplexed stereoscopic images and the selected algorithm may include any of upconversion and/or error correction techniques that contribute to a restoration of the downsampled images.

Abstract: An encoding device evaluates a plurality of processing and/or post-processing algorithms and/or methods to be applied to a video stream, and signals a selected method, algorithm, class or category of methods/algorithms either in an encoded bitstream or as side information related to the encoded bitstream. A decoding device or post-processor utilizes the signaled algorithm or selects an algorithm/method based on the signaled method or algorithm. The selection is based, for example, on availability of the algorithm/method at the decoder/post-processor and/or cost of implementation. The video stream may comprise, for example, downsampled multiplexed stereoscopic images and the selected algorithm may include any of upconversion and/or error correction techniques that contribute to a restoration of the downsampled images.

Abstract: The invention proposes a method and a device for arithmetic encoding of a current spectral coefficient using preceding spectral coefficients. Said preceding spectral coefficients are already encoded and both, said preceding and current spectral coefficients, are comprised in one or more quantized spectra resulting from quantizing time-frequency-transform of video, audio or speech signal sample values.

Abstract: Methods and apparatus for decoding a compressed Higher Order Ambisonics (HOA) representation of a sound or soundfield. The method may include receiving a bit stream containing the compressed HOA representation and decoding, based on a determination that there are multiple layers, the compressed HOA representation from the bitstream to obtain a sequence of decoded HOA representations. A first subset of the sequence of decoded HOA representations is determined based only on corresponding ambient HOA components. A second subset of the sequence of decoded HOA representations is determined based on corresponding ambient HOA components and corresponding predominant sound components.

Abstract: Embodiments are disclosed for timestamp smoothing to remove jitter. In some embodiments, a method of smoothing timestamps associated with audio packets comprises: receiving, using at least one processor, a series of input timestamps for audio packets and their respective packet lengths; estimating, using the at least one processor, an initial timestamp based on the series of input timestamps, the packet lengths and a sample time; calculating, using the at least one processor, a predicted timestamp based on the estimated initial timestamp; and smoothing, using the at least one processor, the predicted timestamp.

Abstract: Methods for generating an object based audio program which is renderable in a personalizable manner, e.g., to provide an immersive, perception of audio content of the program. Other embodiments include steps of delivering (e.g., broadcasting), decoding, and/or rendering such a program. Rendering of audio objects indicated by the program may provide an immersive experience. The audio content of the program may be indicative of multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects, and typically also a default set of objects which will be rendered in the absence of a selection by a user) and a bed of speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Abstract: An encoding device evaluates a plurality of processing and/or post-processing algorithms and/or methods to be applied to a video stream, and signals a selected method, algorithm, class or category of methods/algorithms either in an encoded bitstream or as side information related to the encoded bitstream. A decoding device or post-processor utilizes the signaled algorithm or selects an algorithm/method based on the signaled method or algorithm. The selection is based, for example, on availability of the algorithm/method at the decoder/post-processor and/or cost of implementation. The video stream may comprise, for example, downsampled multiplexed stereoscopic images and the selected algorithm may include any of upconversion and/or error correction techniques that contribute to a restoration of the downsampled images.

Abstract: A quantization parameter signalling mechanism for both SDR and HDR content in video coding is described using two approaches. The first approach is to send the user-defined QpC table directly in high level syntax. This leads to more flexible and efficient QP control for future codec development and video content coding. The second approach is to signal luma and chroma QPs independently. This approach eliminates the need for QpC tables and removes the dependency of chroma quantization parameter on luma QP.

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: In an embodiment, a method comprises: receiving bands of power spectra of an input audio signal and a microphone covariance, and for each band: estimating, using a classifier, respective probabilities of speech and noise; estimating, using a directionality model, a set of means for speech and noise, or a set of means and covariances for speech and noise, based on the microphone covariance for the band and the probabilities; estimating, using a level model, a mean and covariance of noise power based on the probabilities and the power spectra; determining a first noise suppression gain based on the directionality model; determining a second noise suppression gain based on the level model; selecting the first or second noise suppression gain or their sum based on a signal-to-noise ratio of the input audio signal; and scaling a time-frequency representation of the input signal by the selected noise suppression gain.

Abstract: Disclosed is an encoding/decoding method and apparatus related to adaptive deblocking filtering. There is provided an image decoding method performing adaptive filtering in inter-prediction, the method including: reconstructing, from a bitstream, an image signal including a reference block on which block matching is performed in inter-prediction of a current block to be encoded; obtaining, from the bitstream, a flag indicating whether the reference block exists within a current picture where the current block is positioned; reconstructing the current block by using the reference block; adaptively applying an in-loop filter for the reconstructed current block based on the obtained flag; and storing the current block to which the in-loop filter is or is not applied in a decoded picture buffer (DPB).

Abstract: A device and method for video rendering. The device includes a memory and an electronic processor. The electronic processor is configured to receive, from a source device, video data including multiple reference viewpoints, determine a target image plane corresponding to a target viewpoint, determine, within the target image plane, one or more target image regions, and determine, for each target image region, a proxy image region larger than the corresponding target image region. The electronic processor is configured to determine, for each target image region, a plurality of reference pixels that fit within the corresponding proxy image region, project, for each target image region, the plurality of reference pixels that fit within the corresponding proxy image region to the target image region, producing a rendered target region from each target image region, and composite one or more of the rendered target regions to create video rendering.

Abstract: A distributed amplification and packetized audio transmission system for clock synchronization and alignment between an audio/power source and endpoints with dedicated amplifiers and speakers. An Ethernet audio signal is combined with a Power-Line Communications (PLC) signal for transmission from the source to the endpoints over a common conductor. A single master clock in the source synchronizes the Ethernet audio transmitter with the PLC transmitter. Each end-point has a PLC receiver to recover the master clock for use by its Ethernet audio receiver to provide reliable clock synchronization between the source clock and the endpoint clocks. The endpoints can adjust and re-timestamp the PTP packetized clock based upon symbol and timing information from the PLC receiver.

Abstract: Systems, methods, and computer program products for audio processing based on convolutional neural network (CNN) are described. A first CNN architecture may comprise a contracting path of a U-net, a multi-scale CNN, and an expansive path of a U-net. The contracting path may comprise a first encoding layer and may be configured to generate an output representation of the contracting path. The multi-scale CNN may be configured to generate, based on the output representation of the contracting path, an intermediate representation. The multi-scale CNN may comprise at least two parallel convolution paths. The expansive path may comprise a first decoding layer and may be configured to generate a final representation based on the intermediate representation generated by the multi-scale CNN.

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: 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: In an audio encoder, for audio content received in a source audio format, default gains are generated based on a default dynamic range compression (DRC) curve, and non-default gains are generated for a non-default gain profile. Based on the default gains and non-default gains, differential gains are generated. An audio signal comprising the audio content, the default DRC curve, and differential gains is generated. In an audio decoder, the default DRC curve and the differential gains are identified from the audio signal. Default gains are re-generated based on the default DRC curve. Based on the combination of the re-generated default gains and the differential gains, operations are performed on the audio content extracted from the audio signal.

Abstract: Described herein is a method of processing an audio signal using a neural network or using a first and a second neural network. Described is further a method of training said neural network or of jointly training a set of said first and said second neural network. Moreover, described is a method of obtaining and transmitting a latent feature space representation of a perceptual domain audio signal using a neural network and a method of obtaining an audio signal from a latent feature space representation of a perceptual domain audio signal using a neural network. Described are also respective apparatuses and computer program products.