Abstract: A projection system and calibration method therefor relate to a light source configured to emit a light in response to an image data, an illumination optical system configured to steer the light, the illumination optical system including a first lens group and a second lens group, a digital micromirror device (DMD) including a plurality of micromirrors respectively configured to reflect the steered light to a predetermined location as on-state light or to reflect the steered light as off-state light to a light dump; determining a deviation between an actual angle of orientation and an expected angle of orientation of a respective micromirror of the plurality of micromirrors; calculating a first amount of lateral adjustment corresponding to the first lens group and a second amount of lateral adjustment corresponding to the second lens group, and actuating the first and second lens groups according to the corresponding first and second amount.

Abstract: The present invention relates to a method for predicting transform coefficients representing frequency content of an adaptive block length media signal, by receiving a frame and receiving block length information indicating a number of quantized transform coefficients for each block in the frame, the number of quantized transform coefficients being one of a first or second number, wherein the first number is greater than the second number, determining a first block has the second number of quantized transform coefficients, converting the first block into a converted block having the first number of quantized transform coefficients, conditioning a main neural network trained to predict at least one output variable given at least one conditioning variable, the at least one conditioning variable being based on information regarding the converted block and block length information for the first block, providing at least one predicted transform coefficients from an output stage of the main neural network.

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: An audio processing unit (APU) is disclosed. The APU includes a buffer memory configured to store at least one frame of an encoded audio bitstream, where the encoded audio bitstream includes audio data and a metadata container. The metadata container includes a header and one or more metadata payloads after the header. The one or more metadata payloads include dynamic range compression (DRC) metadata, and the DRC metadata is or includes profile metadata indicative of whether the DRC metadata includes dynamic range compression (DRC) control values for use in performing dynamic range compression in accordance with at least one compression profile on audio content indicated by at least one block of the audio data.

Abstract: A system for suppressing noise and enhancing speech and a related method are disclosed. The system trains a neural network model that takes banded energies corresponding to an original noisy waveform and produces a speech value indicating the amount of speech present in each band at each frame. The neural model comprises a feature extraction block that implements some lookahead. The feature extraction block is followed by an encoder with steady down-sampling along the frequency domain forming a contracting path. The encoder is followed by a corresponding decoder with steady up-sampling along the frequency domain forming an expanding path. The decoder receives scaled output feature maps from the encoder at a corresponding level. The decoder is followed by a classification block that generates a speech value indicating an amount of speech present for each frequency band of the plurality of frequency bands at each frame of the plurality of frames.

Abstract: Using a standard-based RGB to YCbCr color transform a new RGB to YCC 3×3 transformation matrix and a 3×1 offset vector are derived under a set of coding-efficiency constraints. The new RGB to YCC 3×3 transform comprises a luminance scaling factor and a 2×2 chroma sub-matrix that preserves the energy of the standard-based RGB to YCbCr transform while maintaining or improving coding efficiency. It also adds support for an authorization or watermarking mechanism in streaming video applications. Examples of using the new color transform using image reshaping are also provided.

Abstract: A global index value is generated for selecting a global reshaping function for an input image of a relatively low dynamic range using luma codewords in the input image. Image filtering is applied to the input image to generate a filtered image. The filtered values of the filtered image provide a measure of local brightness levels in the input image. Local index values are generated for selecting specific local reshaping functions for the input image using the global index value and the filtered values of the filtered image. A reshaped image of a relatively high dynamic range is generated by reshaping the input image with the specific local reshaping functions selected using the local index values.

Abstract: Computer-implemented methods for training a neural network, as well as for implementing audio encoders and decoders via trained neural networks, are provided. The neural network may receive an input audio signal, generate an encoded audio signal and decode the encoded audio signal. A loss function generating module may receive the decoded audio signal and a ground truth audio signal, and may generate a loss function value corresponding to the decoded audio signal. Generating the loss function value may involve applying a psychoacoustic model. The neural network may be trained based on the loss function value. The training may involve updating at least one weight of the neural network.

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 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: A backward reshaping mapping table is initially generated as an inverse of a forward reshaping mapping table. The backward reshaping mapping table is updated by replacing the content-mapped luminance codewords with forward reshaped luminance codewords generated by applying a luminance forward mapping to the sampled luminance codewords. The luminance forward mapping is constructed from the forward reshaping mapping table. The backward reshaping mapping table and the luminance forward mapping are used to generate backward reshaping mappings for creating a reconstructed image from a forward reshaped image. The forward reshaped image is encoded, in a video signal, along with image metadata specifying the backward reshaping mappings. A recipient device of the video signal applies the backward reshaping mappings to the forward reshaped image to create the reconstructed image of the second dynamic range.

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: The present invention relates to a method and device for processing a first and a second audio signal representing an input binaural audio signal acquired by a binaural recording device. The present invention further relates to a method for rendering a binaural audio signal on a speaker system. The method for processing a binaural signal comprising extracting audio information from the first audio signal, computing a band gain for reducing noise in the first audio signal and applying the band gains to respective frequency bands of the first audio signal in accordance with a dynamic scaling factor, to provide a first output audio signal. Wherein the dynamic scaling factor has a value between zero and one and is selected so as to reduce quality degradation for the first audio signal.

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: Media input audio data corresponding to a media stream and microphone input audio data from at least one microphone may be received. A first level of at least one of a plurality of frequency bands of the media input audio data, as well as a second level of at least one of a plurality of frequency bands of the microphone input audio data, may be determined. Media output audio data and microphone output audio data may be produced by adjusting levels of one or more of the first and second plurality of frequency bands based on the perceived loudness of the microphone input audio data, of the microphone output audio data, of the media output audio data and the media input audio data. One or more processes may be modified upon receipt of a mode-switching indication.

Abstract: The quantization parameter QP is well-known in digital video compression as an indication of picture quality. Digital symbols representing a moving image are quantized with a quantizing step that is a function QSN of the quantization parameter QP, which function QSN has been normalized to the most significant bit of the bit depth of the digital symbols. As a result, the effect of a given QP is essentially independent of bit depth a particular QP value has a standard effect on image quality, regardless of bit depth. The invention is useful, for example, in encoding and decoding at different bit depths, to generate compatible, bitstreams having different bit depths, and to allow different bit depths for different components of a video signal by compressing each with the same fidelity (i.e., the same QP).

Abstract: Several embodiments of scalable image processing systems and methods are disclosed herein whereby color management processing of source image data to be displayed on a target display is changed according to varying levels of metadata.

Abstract: Diffuse or spatially large audio objects may be identified for special processing. A decorrelation process may be performed on audio signals corresponding to the large audio objects to produce decorrelated large audio object audio signals. These decorrelated large audio object audio signals may be associated with object locations, which may be stationary or time-varying locations. For example, the decorrelated large audio object audio signals may be rendered to virtual or actual speaker locations. The output of such a rendering process may be input to a scene simplification process. The decorrelation, associating and/or scene simplification processes may be performed prior to a process of encoding the audio data.

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.