Abstract: One or more perforation hole pattern methods are applied (402) to generate a spatial distribution of perforation holes forming a semi-random pattern for an image display screen. The image display screen is perforated (404) with the spatial distribution of perforation holes forming the semi-random pattern. Image rendering light is emitted (406) with a light projector toward the image display screen that is installed in an image rendering environment. At least a portion of the image rendering light emitted from the light projector is reflected (408) by the image display screen, toward a viewer.

Abstract: The present document relates an audio encoding and decoding system (referred to as an audio codec system). In particular, the present document relates to a audio codec system which is particularly well suited for voice encoding/decoding. A transform-based speech encoder is configured to encode a speech signal into a bitstream is described. A speech decoder configured to decode audio signals from a bitstream is further described.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.

Abstract: Methods and audio processing units for generating an object based audio program including conditional rendering metadata corresponding to at least one object channel of the program, where the conditional rendering metadata is indicative of at least one rendering constraint, based on playback speaker array configuration, which applies to each corresponding object channel, and methods for rendering audio content determined by such a program, including by rendering content of at least one audio channel of the program in a manner compliant with each applicable rendering constraint in response to at least some of the conditional rendering metadata. Rendering of a selected mix of content of the program may provide an immersive experience.

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 are described for processing an image captured with an image sensor, such as a camera. In one embodiment, an estimated ambient light level of the captured image is determined and used to compute an optical-optical transfer function (OOTF) that is used to correct the image to preserve an apparent contrast of the image under the estimated ambient light level in a viewing environment. The estimated ambient light level is determined by scaling pixel values from the image sensor using a function that includes exposure parameters and a camera specific parameter derived from a camera calibration.

Abstract: Volume leveler controller and controlling method are disclosed. In one embodiment, A volume leveler controller includes an audio content classifier for identifying the content type of an audio signal in real time; and an adjusting unit for adjusting a volume leveler in a continuous manner based on the content type as identified. The adjusting unit may configured to positively correlate the dynamic gain of the volume leveler with informative content types of the audio signal, and negatively correlate the dynamic gain of the volume leveler with interfering content types of the audio signal.

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: Embodiments are disclosed for projection systems with rotatable anamorphic lenses. In an embodiment, an optical projection system comprises: a light source; an optical integrator configured to receive light from the light source and to distribute a uniform pattern of light; a relay lens system including two or more rotatable anamorphic lenses, the anamorphic lenses oriented about an optical axis to transform the uniform pattern of light into an image having a specified aspect ratio; at least one spatial light modulator configured to receive the image and direct a spatially modulated image along an optical path; and at least one projection lens configured to receive the spatially modulated image from the optical path and to project the spatially modulated image onto an image plane with the specified aspect ratio.

Abstract: In some embodiments, a method for performing at least one of enhancement, decoding, or rendering of a multichannel audio signal in response to compression feedback or feedback from a smart amplifier. For example, the compression feedback may be indicative of amount of compression applied to each of multiple frequency bands, of the audio signal or an enhanced audio signal generated in response thereto. The enhancement (e.g., bass enhancement) may include dynamic routing of audio content of the input audio signal between channels of an enhanced audio signal generated in response thereto. The enhancement and compression may be performed on a per speaker class basis. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook or laptop computer, tablet, soundbar, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: A method for distributing High Dynamic Range (HDR) content to playback devices for displaying images where the HDR content is encoded to an HDR bitstream and the HDR bitstream is subsequently decoded by a playback device. The HDR bitstream contains auxiliary metadata packets that are based upon the processing capability of the playback device.

Abstract: A computer-implemented method of audio processing, the method comprising: receiving audio object data and audio description data, wherein the audio object data includes a first plurality of audio objects; calculating a long-term loudness of the audio object data and a long- term loudness of the audio description data; calculating a plurality of short-term loudnesses of the audio object data and a plurality of short-term loudnesses of the audio description data; reading a first plurality of mixing parameters that correspond to the audio object data; generating a second plurality of mixing parameters based on the first plurality of mixing parameters, the long-term loudness of the audio object data, the long-term loudness of the audio description data, the plurality of short-term loudnesses of the audio object data, and the plurality of short-term loudnesses of the audio description data; generating a gain adjustment visualization corresponding to the second plurality of mixing parameters, the audio object data

Abstract: Some methods involve receiving an input audio signal that includes N input audio channels, the input audio signal representing a first soundfield format having a first soundfield format resolution, N being an integer ?2. A first decorrelation process may be applied to two or more of the input audio channels to produce a first set of decorrelated channels, the first decorrelation process maintaining an inter-channel correlation of the set of input audio channels. A first modulation process may be applied to the first set of decorrelated channels to produce a first set of decorrelated and modulated output channels. The first set of decorrelated and modulated output channels may be combined with two or more undecorrelated output channels to produce an output audio signal that includes O output audio channels representing a second and relatively higher-resolution soundfield format than the first soundfield format, O being an integer ?3.

Abstract: A content-creation tool includes a processor and a memory. The processor is configured to receive a first video clip and a second video clip, a respective first and second metadata-item thereof being set to a respective first and second metadata-value. The memory stores video-editing software that includes a timeline interface and instructions that, when executed by the processor, control the processor to: add the first video clip to the timeline interface as a first timeline-track that retains the first metadata-value; add the second video clip to the timeline interface as a second timeline-track that retains the second metadata-value; and generate a frame sequence that includes a plurality of video frames. Each video frame is a frame of, or a frame derived from, one of (i) the first timeline-track, (ii) the second timeline-track, and (iii) a composited time-line-track composited from at least one of the first and second timeline-tracks.

Abstract: Embodiments are disclosed for non-intrusive transducer health detection in an audio system. In an embodiment, a method performed by the audio system comprises outputting one or more encoded inaudible acoustic signals into an acoustic transmission medium using a first transducer. The one or more encoded inaudible acoustic signals are received from the acoustic transmission medium using a second transducer of the audio system. The received one or more encoded inaudible acoustic signals are used to identify failure or degradation of the first or second transducer.

Abstract: A method comprises: obtaining softmask values for frequency bins of time-frequency tiles representing an audio signal; reducing, or expanding and limiting, the softmask values; and applying the reduced, or expanded and limited, softmask values to the frequency bins to create a time-frequency representation of an estimated target source. An alternative method comprises, for each time-frequency tile: obtaining softmask values; applying the softmask values to the frequency bins to create a time-frequency domain representation of an estimated target source; obtaining a panning parameter and a source concentration estimates for the target source; determining, using the panning parameter estimate and the softmask values, a magnitude for the time-frequency representation of the estimated target source; determining, using the panning parameter estimate and the source phase concentration estimate, a phase for the time-frequency representation of the estimated target source; and combining the magnitude and the phase.

Abstract: Described herein is a method of determining parameters for a generative neural network for processing an audio signal, wherein the generative neural network includes an encoder stage mapping to a coded feature space and a decoder stage, each stage including a plurality of convolutional layers with one or more weight coefficients, the method comprising a plurality of cycles with sequential processes of: pruning the weight coefficients of either or both stages based on pruning control information, the pruning control information determining the number of weight coefficients that are pruned for respective convolutional layers; training the pruned generative neural network based on a set of training data; determining a loss for the trained and pruned generative neural network based on a loss function; and determining updated pruning control information based on the determined loss and a target loss. Further described are corresponding apparatus, programs, and computer-readable storage media.

Abstract: A system and method of editing video content includes receiving input video data; converting the input video data to a predetermined format; generating a plurality of initial metadata values for a frame of the converted video data, the plurality of initial metadata values including a first metadata value corresponding to a first fixed value not calculated from a content including the frame, a second metadata value corresponding to an average luminance value of the frame, and a third metadata value corresponding to a second fixed value not calculated from the content, wherein the first meta-data value, the second metadata value, and the third metadata value include information used by a decoder to render a decoded image on a display.

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: Disclosed are methods and systems which convert a multi-microphone input signal to a multichannel output signal making use of a time- and frequency-varying matrix. For each time and frequency tile, the matrix is derived as a function of a dominant direction of arrival and a steering strength parameter. Likewise, the dominant direction and steering strength parameter are derived from characteristics of the multi-microphone signals, where those characteristics include values representative of the inter-channel amplitude and group-delay differences.

Abstract: A system utilizing a high throughput coding mode for CABAC in HEVC is described. The system may include an electronic device configured to obtain a block of data to be encoded using an arithmetic based encoder; to generate a sequence of syntax elements using the obtained block; to compare an Absolute-3 value of the sequence or a parameter associated with the Absolute-3 value to a preset value; and to convert the Absolute-3 value to a codeword using a first code or a second code that is different than the first code, according to a result of the comparison.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M?N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

Abstract: Based on viewing tracking data, a viewer's view direction to a three-dimensional (3D) scene depicted by a first video image is determined. The first video image has been streamed in a video stream to the streaming client device before the first time point and rendered with the streaming client device to the viewer at the first time point. Based on the viewer's view direction, a target view portion is identified in a second video image to be streamed in the video stream to the streaming client device to be rendered at a second time point subsequent to the first time point. The target view portion is encoded into the video stream with a higher target spatiotemporal resolution than that used to encode remaining non-target view portions in the second video image.

Abstract: Occluded image fragments are sorted in size. The largest image fragment is used to size a quadtree node in a layout mask for a disocclusion atlas used to store the image fragments. The sorted image fragments are stored into the disocclusion atlas using the layout mask such as each image fragment is hosted with a best fit quadtree node in the disocclusion atlas. A video signal may be generated by encoding one or more reference images and the disocclusion atlas storing the image fragments. The image fragments can be used by a recipient device to fill disoccluded image data in disoccluded spatial regions in a display image synthesized from the reference 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: 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 present document describes a method (700) for encoding a multi-channel input signal (201). The method (700) comprises determining (701) a plurality of downmix channel signals (203) from the multi-channel input signal (201) and performing (702) energy compaction of the plurality of downmix channel signals (203) to provide a plurality of compacted channel signals (404). Furthermore, the method (700) comprises determining (703) joint coding metadata (205) based on the plurality of compacted channel signals (404) and based on the multi-channel input signal (201), wherein the joint coding metadata (205) is such that it allows upmixing of the plurality of compacted channel signals (404) to an approximation of the multi-channel input signal (201). In addition, the method (700) comprises encoding (704) the plurality of compacted channel signals (404) and the joint coding metadata (205).

Abstract: Projection displays include a highlight projector and a main projector. Highlights projected by the highlight projector boost luminance in highlight areas of a base image projected by the main projector. Various highlight projectors including steerable beams, holographic projectors and spatial light modulators are described.

Abstract: The described embodiments include systems and methods for producing and adapting images, such as video images, for presentation on display devices that have various different aspects ratios, such as 4:3, 16:9, 9:16, etc. In one embodiment, a method for producing content, such as video images, can begin by selecting an original aspect ratio and determining, within at least a first scene in the content, a position of a subject in the first scene. In one embodiment, the original aspect ratio can be substantially square (e.g., 1:1). Metadata can then be created, based on the position of the subject in the first scene, to guide playback devices to asymmetrically crop the content, relative to the position, for display on display devices that have aspect ratios that are different than the original aspect ratio. Other methods and systems are also described.

Abstract: The present document describes a method (600) for encoding a multi-channel input signal (101) which comprises N different channels. The method (600) comprises, for a current frame of a sequence of frames, determining (601) whether the current frame is an active frame or an inactive frame using a signal and/or a voice activity detector, and determining (602) a downmix signal (103) based on the multi-channel input signal (101), wherein the downmix signal (103) comprises N channels or less. In addition, the method (600) comprises determining (603) upmixing metadata (105) comprising a set of parameters for generating, based on the downmix signal (103), a reconstructed multi-channel signal (111) comprising N channels, wherein the upmixing metadata (105) is determined in dependence of whether the current frame is an active frame or an inactive frame. The method (600) further comprises encoding (604) the upmixing metadata (105) into a bitstream.

Abstract: Saliency regions are identified in a global scene depicted by volumetric video. Saliency video streams that track the saliency regions are generated. Each saliency video stream tracks a respective saliency region. A saliency stream based representation of the volumetric video is generated to include the saliency video streams. The saliency stream based representation of the volumetric video is transmitted to a video streaming client.

Abstract: Sampled data is packaged in checkerboard format for encoding and decoding. The sampled data may be quincunx sampled multi-image video data (e.g., 3D video or a multi-program stream), and the data may also be divided into sub-images of each image which are then multiplexed, or interleaved, in frames of a video stream to be encoded and then decoded using a standardized video encoder. A system for viewing may utilize a standard video decoder and a formatting device that de-interleaves the decoded sub-images of each frame reformats the images for a display device. A 3D video may be encoded using a most advantageous interleaving format such that a preferred quality and compression ratio is reached. In one embodiment, the invention includes a display device that accepts data in multiple formats.

Abstract: Methods and systems for performing at least one audio activity (e.g., conducting a phone call or playing music or other audio content) in an environment including by determining an estimated location of a user in the environment in response to sound uttered by the user (e.g., a voice command), and controlling the audio activity in response to determining the estimated user location. The environment may have zones which are indicated by a zone map and estimation of the user location may include estimating in which of the zones the user is located. The audio activity may be performed using microphones and loudspeakers which are implemented in or coupled to smart audio devices.

Abstract: The present document relates to a method of layered encoding of a compressed sound representation of a sound or sound field. The compressed sound representation comprises a basic compressed sound representation comprising a plurality of components, basic side information for decoding the basic compressed sound representation to a basic reconstructed sound representation of the sound or sound field, and enhancement side information including parameters for improving the basic reconstructed sound representation.

Abstract: A method of audio processing includes generating harmonics in a hybrid complex quadrature mirror filter domain. Generating the harmonics may include multiplication, using a feedback delay loop, and dynamic compression. The harmonics may be generated based on one or more hybrid sub-bands of the complex transform domain signal.

Abstract: Computer-implemented methods and devices for combined audio separation and classification are provided. An estimated separated signal is time gated based on a determination of an audio classifier of, at least in part, the original mix of signals before separation. Combined separation, classification, and time gating of both the estimated signal and a residual signal are also provided.

Abstract: Systems, methods, and computer program products are disclosed for adaptive downmixing of audio signals with improved continuity. An audio encoding system receives an input multi-channel audio signal including a primary input audio channel and L non-primary input audio channels. The system determines a set of L input gains. For each of the channels and gains, the system forms a respective scaled non-primary input audio channel. The system forms a primary output audio channel from the sum of the primary input audio channel and the scaled non-primary input audio channels. The system determines a set of L prediction gains. The system forms a prediction channel from the primary output audio channel. The system forms L non-primary output audio channels. The system forms an output multi-channel audio signal from the primary output audio channel and the L non-primary output audio channels.

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

Abstract: While a viewer is viewing a first stereoscopic image comprising a first left image and a first right image, a left vergence angle of a left eye of a viewer and a right vergence angle of a right eye of the viewer are determined. A virtual object depth is determined based at least in part on (i) the left vergence angle of the left eye of the viewer and (ii) the right vergence angle of the right eye of the viewer. A second stereoscopic image comprising a second left image and a second right image for the viewer is rendered on one or more image displays. The second stereoscopic image is subsequent to the first stereoscopic image. The second stereoscopic image is projected from the one or more image displays to a virtual object plane at the virtual object depth.

Abstract: An apparatus and method relating to use of a physical writing surface (132) during a videoconference or presentation. Snapshots of a whiteboard (132) are identified by applying a difference measure to the video data (e.g., as a way of comparing frames at different times). Audio captured by a microphone may be processed to generate textual data, wherein a portion of the textual data is associated with each snapshot. The writing surface may be identified (enrolled) using gestures. Image processing techniques may be used to transform views of a writing surface.

Abstract: An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

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: 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: 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: An audio session management method may involve: determining, by an audio session manager, one or more first media engine capabilities of a first media engine of a first smart audio device, the first media engine being configured for managing one or more audio media streams received by the first smart audio device and for performing first smart audio device signal processing for the one or more audio media streams according to a first media engine sample clock; receiving, by the audio session manager and via a first application communication link, first application control signals from the first application; and controlling the first smart audio device according to the first media engine capabilities, by the audio session manager, via first audio session management control signals transmitted to the first smart audio device via a first smart audio device communication link and without reference to the first media engine sample clock.

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: A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

Abstract: An intra prediction mode encoding and decoding method, an image decoding device, and an image encoding device operate by deriving most probable modes (MPMs) from surrounding prediction units adjacent to a current prediction unit and deriving an intra prediction mode of the current prediction unit on the basis of an MPM flag indicating whether an MPM having the same prediction mode as the intra prediction mode of the current prediction unit exists among the derived MPMs.

Abstract: An integrated cinema amplifier comprises a power supply stage that distributes power over a plurality of channels for rendering immersive audio content in a surround sound listening environment. The amplifier automatically detects maximum and net power availability and requirements based on audio content by decoding audio metadata and dynamically adjusts gains to each channel or sets of channels based on content and operational/environmental conditions. A power supply stage provides power to drive a plurality of channels corresponding to speaker feeds to a plurality of speakers. The amplifier has a front panel having an LED array with each LED associated with a respective channel or group of channels of the multi-channel amplifier, and a control unit configured to light the LEDs according to display patterns based on operating status or error conditions of the amplifier.

Abstract: Methods, systems, and computer program products for rending an audio object having an apparent size are disclosed. An audio processing system receives audio panning data including a first grid mapping first virtual sound sources in a space and speaker positions to speaker gains. The first grid specifies first speaker gains of the first virtual sound sources in the space. The audio processing system determines a second grid of second virtual sound sources in the space, including mapping the first virtual sound sources into the second virtual sound sources of the second virtual sources. The audio processing system selects at least one of the first grid or second grid for rendering an audio object based on an apparent size of the audio object. The audio processing system renders the audio object based on the selected grid or grids.