Abstract: On the basis of a bitstream (P), an n-channel audio signal (X) is reconstructed by deriving an m-channel core signal (Y) and multichannel coding parameters (?) from the bitstream, where 1?m<n. Also derived from the bitstream are pre-processing dynamic range control, DRC, parameters (DRC2) quantifying an encoder-side dynamic range limiting of the core signal. The n-channel audio signal is obtained by parametric synthesis in accordance with the multichannel coding parameters and while cancelling any encoder-side dynamic range limiting based on the pre-processing DRC parameters. In particular embodiments, the reconstruction further includes use of compensated post-processing DRC parameters quantifying a potential decoder-side dynamic range compression. Cancellation of an encoder-side range limitation and range compression are preferably performed by different decoder-side components. Cancellation and compression may be coordinated by a DRC pre-processor.

Abstract: A novel light source for a 3D display system includes a plurality of left eye light emitters and a plurality of right eye light emitters. The left eye emitters include a broad spectral distribution emitter and an overlapping narrow spectral distribution emitter in each of the blue, green, and red color bands. Similarly, the right eye emitters include a broad spectral distribution emitter and an overlapping narrow spectral distribution emitter in each of the blue, green, and red color bands. The combined spectral distributions of each of the broad and narrow emitters provide a primary light for each color and for each eye that has a desirable spectral shape, including wide bandwidth and short tail(s). The invention thus minimizes cross-talk and speckling in left- and right-eye images of 3D display systems.

Abstract: One or more media contents are received. A viewer's light adaptive states are predicted as a function of time as if the viewer is watching display mapped images derived from the one or more media contents. The viewer's light adaptive states are used to detect an excessive change in luminance in a specific media content portion of the one or more media contents. The excessive change in luminance in the specific media content portion of the one or more media contents is caused to be reduced while the viewer is watching one or more corresponding display mapped images derived from the specific media content portion of the one or more media contents.

Abstract: Input video data of a video dynamic range and its input dynamic metadata is received. Input graphics data of a graphics dynamic range is received. Display identification data is received from a target display over a video interface. Interpolated dynamic metadata is generated based at least in part on (a) the input dynamic metadata, (b) the display identification data, and (c) a numeric interpolation factor, in order to operate a priority-transition mode for transitioning between a video-priority mode and a graphics-priority mode. The input video data is blended with the input graphics data based at least in part on the interpolated dynamic metadata into graphics blended video data. The graphics blended video data and the interpolated dynamic metadata is sent to the target display for rendering graphics blended images represented in the graphics blended video data.

Abstract: A method for encoding a LUT defined as a lattice of vertices is disclosed. At least one value is of each vertex of the lattice. The method comprises for a current vertex: predicting the at least one value of said current vertex from another value which is for example obtained from reconstructed values of neighboring vertices; and encoding in a bitstream at least one residue computed between the at least one value of the current vertex and its prediction in a bitstream.

Abstract: Dual and multi-modulator projector display systems and techniques are disclosed. In one embodiment, a projector display system comprises a light source; a controller, a first modulator, receiving light from the light source and rendering a halftone image of said the input image; a blurring optical system that blurs said halftone image with a Point Spread Function (PSF); and a second modulator receiving the blurred halftone image and rendering a pulse width modulated image which may be projected to form the desired screen image. Systems and techniques for forming a binary halftone image from input image, correcting for misalignment between the first and second modulators and calibrating the projector system—e.g. over time—for continuous image improvement are also disclosed.

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: 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/?{square root over (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: Methods, systems, and computer program products that provide streaming capabilities to audio input and output devices are disclosed. An audio processing device connects an upstream device to a downstream device. The upstream device is plugged into an input port of the audio processing device. The audio processing device intercepts a signal from the upstream device to the downstream device. The audio processing device converts the signal to digital data and streams the digital data to a server. The digital data can include metadata, e.g., an input gain. The audio processing device can adjust the input gain in response to instructions from the server. The audio processing device feeds a pass-through copy of the audio signal to an output port. A user can connect the downstream device in a usual signal chain into the output port of the audio processing device. The streaming does not affect the user's workflow.

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: Distortion reducing multi-band compressor with timbre preservation is provided. Timbre preservation is achieved by determining a time-varying threshold in each of a plurality frequency bands as a function of a respective fixed threshold for the frequency band and, at least in part, an audio signal level and a fixed threshold outside such frequency band. If a particular frequency band receives significant gain reduction due to being above or approaching its fixed threshold, then a time-varying threshold of one or more other frequency bands are also decreased to receive some gain reduction. In a specific embodiment, time-varying thresholds can be computed from an average difference of the audio input signal in each frequency band and its respective fixed threshold.

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: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A compression process reduces an original dynamic range of an initial audio signal through a compression process that divides the initial audio signal into a plurality of segments using a defined window shape, calculates a wideband gain in the frequency domain using a non-energy based average of frequency domain samples of the initial audio signal, and applies individual gain values to amplify segments of relatively low intensity and attenuate segments of relatively high intensity. The compressed audio signal is then expanded back to the substantially the original dynamic range that applies inverse gain values to amplify segments of relatively high intensity and attenuating segments of relatively low intensity. A QMF filterbank is used to analyze the initial audio signal to obtain a frequency domain representation.

Abstract: Systems and methods are described for detecting and remedying potential incongruence in a video conference. A camera of a video conferencing system may capture video images of a conference room. A processor of the video conferencing system may identify locations of a plurality of participants within an image plane of a video image. Using face and shape detection, a location of a center point of each identified participant's torso may be calculated. A region of congruence bounded by key parallax lines may be calculated, the key parallax lines being a subset of all parallax lines running through the center points of each identified participant. When the audio device location is not within the region of congruence, audio captured by an audio device may be adjusted to reduce effects of incongruence when the captured audio is replayed at a far end of the video conference.

Abstract: Many portable playback devices cannot decode and playback encoded audio content having wide bandwidth and wide dynamic range with consistent loudness and intelligibility unless the encoded audio content has been prepared specially for these devices. This problem can be overcome by including with the encoded content some metadata that specifies a suitable dynamic range compression profile by either absolute values or differential values relative to another known compression profile. A playback device may also adaptively apply gain and limiting to the playback audio. Implementations in encoders, in transcoders and in decoders are disclosed.

Abstract: A method for encoding a LUT defined as a lattice of vertices is disclosed. At least one value is of each vertex of the lattice. The method comprises for a current vertex: predicting the at least one value of said current vertex from another value which is for example obtained from reconstructed values of neighboring vertices; and encoding in a bitstream at least one residue computed between the at least one value of the current vertex and its prediction in a bitstream.

Abstract: The present disclosure relates to a method of converting an audio signal in an intermediate signal format to a set of speaker feeds suitable for playback by an array of speakers. The audio signal in the intermediate signal format is obtainable from an input audio signal by means of a spatial panning function. The method comprises determining a discrete panning function for the array of speakers, determining a target panning function based on the discrete panning function, wherein determining the target panning function involves smoothing the discrete panning function, and determining a rendering operation for converting the audio signal in the intermediate signal format to the set of speaker feeds, based on the target panning function and the spatial panning function. The present disclosure further relates to a corresponding apparatus and a corresponding computer-readable storage medium.

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: Embodiments are directed to a method and system for receiving, in a bitstream, metadata associated with the audio data, and analyzing the metadata to determine whether a loudness parameter for a first group of audio playback devices are available in the bitstream. Responsive to determining that the parameters are present for the first group, the system uses the parameters and audio data to render audio. Responsive to determining that the loudness parameters are not present for the first group, the system analyzes one or more characteristics of the first group, and determines the parameter based on the one or more characteristics.