Abstract: The present document relates to audio source coding systems. In particular, the present document relates to audio source coding systems which make use of linear prediction in combination with a filterbank. A method for estimating a first sample (615) of a first subband signal in a first subband of an audio signal is described. The first subband signal of the audio signal is determined using an analysis filterbank (612) comprising a plurality of analysis filters which provide a plurality of subband signals in a plurality of subbands from the audio signal, respectively.

Abstract: The present disclosure relates to a method of downmixing a plurality of input audio channels. The method include obtaining, for each of the input audio channels, a plurality of frequency coefficients in a plurality of corresponding frequency bins, and applying, for at least one frequency bin, a downmix matrix to a first array formed by the frequency coefficients of the plurality of input audio channels for the respective frequency bin to obtain a second array formed by the frequency coefficients of a plurality of intermediate audio channels for the respective frequency bin. The method further involves determining a third array including only the non-zero entries of the downmix matrix, and determining a fourth array including, for each entry of the third array, an entry indicative of a position of the respective entry of the third array within the downmix matrix.

Abstract: A system for decoding a video bitstream includes receiving a frame of the video that includes at least one slice and at least one tile and where each of the at least one slice and the at least one tile are not all aligned with one another.

Abstract: There is provided a method for enhancing dialog in a decoder of an audio system. The method comprises receiving a plurality of downmix signals being a downmix of a larger plurality of channels; receiving parameters for dialog enhancement being defined with respect to a subset of the plurality of channels that is downmixed into a subset of the plurality of downmix signals; upmixing the subset of downmix signals parametrically in order to reconstruct the subset of the plurality of channels with respect to which the parameters for dialog enhancement are defined; applying dialog enhancement to the subset of the plurality of channels with respect to which the parameters for dialog enhancement are defined using the parameters for dialog enhancement to provide at least one dialog enhanced signal; and subjecting the at least one dialog enhanced signal to mixing to provide dialog enhanced versions of the subset of downmix signals.

Abstract: An audio decoding system (100) for processing a two-channel input signal (X) comprises a parametric mixing stage (110). The parametric mixing stage receives the two-channel input signal and a set of mixing parameters (P1), and outputs a two-channel output signal (Y1). The parametric mixing stage comprises a decorrelation stage (111) outputting a decorrelated signal (D1) based on the input signal. The parametric mixing stage further comprises a mixing matrix (112) receiving the input signal and the de-correlated signal, and forming a two-channel linear combination of channels from the input signal and the decorrelated signal. The mixing matrix outputs the linear combination as the two-channel output signal. Coefficients of the linear combination are controllable by the set of mixing parameters, and at least four mixing parameters of the set are independently assignable.

Abstract: Stereoscopic video data and corresponding depth map data for stereoscopic and auto-stereoscopic displays are coded using a coded base layer and one or more coded enhancement layers. Given a 3D input picture and corresponding input depth map data, a side-by-side and a top-and-bottom picture are generated based on the input picture. Using an encoder, the side-by-side picture is coded to generate a coded base layer Using the encoder and a texture reference processing unit (RPU), the top-and-bottom picture is encoded to generate a first enhancement layer, wherein the first enhancement layer is coded based on the base layer stream, and using the encoder and a depth-map RPU, depth data for the side-by-side picture are encoded to generate a second enhancement layer, wherein the second enhancement layer is coded based on to the base layer. Alternative single, dual, and multi-layer depth map delivery systems are also presented.

Abstract: This disclosure falls into the field of audio coding, in particular it is related to the field of spatial audio coding, where the audio information is represented by multiple audio objects including at least one dialog object. In particular the disclosure provides a method and apparatus for enhancing dialog in a decoder in an audio system. Furthermore, this disclosure provides a method and apparatus for encoding such audio objects for allowing dialog to be enhanced by the decoder in the audio system.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding a stereo audio signal based on an input signal. According to the disclosure, a hybrid approach of using both parametric stereo coding and a discrete representation of the stereo audio signal is used which may improve the quality of the encoded and decoded audio for certain bitrates.

Abstract: Encoding and decoding devices for encoding the channels of an audio system having at least four channels are disclosed. The decoding device has a first stereo decoding component which subjects a first pair of input channels to a first stereo decoding, and a second stereo decoding component which subjects a second pair of input channels to a second stereo decoding. The results of the first and second stereo decoding components are crosswise coupled to a third and a fourth stereo decoding component which each performs stereo decoding on one channel resulting from the first stereo decoding component, and one channel resulting from the second stereo decoding component.

Abstract: The present invention proposes a new method for improving the performance of a real-valued filterbank based spectral envelope adjuster. By adaptively locking the gain values for adjacent channels dependent on the sign of the channels, as defined in the application, reduced aliasing is achieved. Furthermore, the grouping of the channels during gain-calculation, gives an improved energy estimate of the real valued subband signals in the filterbank.

Abstract: Embodiments are directed to a companding method and system for reducing coding noise in an audio codec. A method of processing an audio signal comprises receiving an audio signal, classifying the audio signal as one of pure sinusoidal, hybrid, or pure transient signal using two defined threshold values, and selectively applying a companding operation by switching between a companding off mode, a companding on mode, and an average companding mode, comprising selecting between the companding on mode and the average companding mode for a classified hybrid signal using a companding rule that uses a temporal sharpness measure in a frequency domain.

Abstract: A method of encoding channel or object based input audio for playback, the method including the steps of: (a) initially rendering the channel or object based input audio into an initial output presentation; (b) determining an estimate of the dominant audio component from the channel or object based input audio and determining a series of dominant audio component weighting factors for mapping the initial output presentation into the dominant audio component; (c) determining an estimate of the dominant audio component direction or position; and (d) encoding the initial output presentation, the dominant audio component weighting factors, the dominant audio component direction or position as the encoded signal for playback.

Abstract: A spatial adaptation system for multiple-microphone sound capture systems and methods thereof are described. A spatial adaptation system includes an inference and weight module configured to receive a inputs. The inputs based on two or more input signals captured by at least two microphones. The inference and weight module to determine one or more weight values base on at least one of the inputs. The spatial adaptation system also including a noise magnitude ratio update module coupled with the inference and weight module. The noise magnitude ratio update module to determine an updated noise target based on the one or more weight values from the inference and weight module.

Abstract: The positions of a plurality of speakers at a media consumption site are determined. Audio information in an object-based format is received. Gain adjustment value for a sound content portion in the object-based format may be determined based on the position of the sound content portion and the positions of the plurality of speakers. Audio information in a ring-based channel format is received. Gain adjustment value for each ring-based channel in a set of ring-based channels may be determined based on the ring to which the ring-based channel belongs and the positions of the speakers at a media consumption site.

Abstract: The present disclosure provides methods, devices and computer program products for non-uniform quantization of parameters relating to parametric spatial coding of audio signals. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system taking the non-uniformly quantized parameters into account. According to the disclosure, such an approach renders it possible to reduce bit consumption without substantially reducing the quality of the reconstructed audio object.

Abstract: A method for hybrid speech enhancement which employs parametric-coded enhancement (or blend of parametric-coded and waveform-coded enhancement) under some signal conditions and waveform-coded enhancement (or a different blend of parametric-coded and waveform-coded enhancement) under other signal conditions. Other aspects are methods for generating a bitstream indicative of an audio program including speech and other content, such that hybrid speech enhancement can be performed on the program, a decoder including a buffer which stores at least one segment of an encoded audio bitstream generated by any embodiment of the inventive method, and a system or device (e.g., an encoder or decoder) configured (e.g., programmed) to perform any embodiment of the inventive method. At least some of speech enhancement operations are performed by a recipient audio decoder with Mid/Side speech enhancement metadata generated by an upstream audio encoder.

Abstract: A method for encoding an image having been cut up into partitions. The method includes: predicting data of a current partition based on an already encoded and then decoded reference partition, generating a predicted partition; determining residual data by comparing data relating to the current partition with the predicted partition, the residual data associated with various digital data items. Prior producing a signal containing the encoded information, performing the following steps; determining, from the predetermined residual data, a subset containing residual data capable of being modified; calculating the value of a function representative of the residual data; comparing the calculated value with a value of at least one of the digital data items; based on the comparison, modification or non-modification of at least one of the residual data items of the subset; and, in the event of a modification, entropy encoding the at least one modified residual data item.

Abstract: Techniques for low bit rate parametric encoding of haptic-tactile signals. The techniques encompass a parametric encoding method. The parametric encoding method includes the steps of: for at least one frame of a plurality of frames of a source haptic-tactile signal, representing the source haptic-tactile signal in the frame as a set of parameters and according to a functional representation; and including the set of parameters in a bit stream that encodes the source haptic-tactile signal. The functional representation is based on one of a set of orthogonal functionals, or polynomial approximation. For example, the functional representation can be based on one of Chebyshev functionals of the first kind through order n, Chebyshev functionals of the second kind through order n, or k-th order polynomial approximation.

Abstract: Methods and systems for generating an audio mix indicative of action sound captured at an event on a surface (e.g., a sporting event on a field) using a microphone array, where the array includes subsurface microphones (e.g., a large number of subsurface microphones) positioned under the surface, and optionally also other microphones. In typical embodiments, at least one point of interest (“PI”) on the surface is selected in an automated manner, PI data indicative of a currently selected PI on the surface is generated (e.g., a sequence of PIs on the surface is selected, the PI data is indicative of the sequence of PIs, and a most recently selected PI in the sequence is the currently selected PI), and the audio mix is generated in response to the PI data. Aspects include methods performed by any embodiment of the system, and a system or device configured (e.g., programmed) to perform any embodiment of the method.

Abstract: Embodiments relate to an audio processing unit that includes a buffer, bitstream payload deformatter, and a decoding subsystem. The buffer stores at least one block of an encoded audio bitstream. The block includes a fill element that begins with an identifier followed by fill data. The fill data includes at least one flag identifying whether enhanced spectral band replication (eSBR) processing is to be performed on audio content of the block. A corresponding method for decoding an encoded audio bitstream is also provided.