The present disclosure provides methods, devices and computer program products which provide less complex and more flexible control of the introduced decorrelation in an audio coding system. According to the disclosure, this is achieved by calculating and using two weighting factors, one for an approximated audio object and one for a decorrelated audio object, for introduction of decorrelation of audio objects in the audio coding system.

A system and method for rendering a piece of sound program content to include data and parameters that describe perceptual, acoustic, and geometric object properties is provided. The perceptual, acoustic, and geometric properties may include one or more of 1) a three-dimensional location of an audio object, 2) a width of an audio object, 3) ambience characteristics of an audio object, 4) diffuseness characteristics of an audio object, and 5) a direct-to-reverberant sound ratio of an audio object. Based on these pieces of data, an audio playback system may produce one or more beam patterns that reproduce three-dimensional properties of audio objects and/or audio channels of the piece of sound program content. Accordingly, the system and method for rendering a piece of sound program content may accurately represent the multi-dimensional properties of the piece of sound program content through the use of beam patterns.

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.

An apparatus comprising means for: receiving values for sub-bands of a frame of an audio signal, the values comprising at least one azimuth value, at least one elevation value at least one energy ratio value and at least one spread and/or surround coherence value for each sub-band; determining a codebook for encoding at least one spread and/or surround coherence value for each sub-band based on the at least one energy ratio value and at least one azimuth value for each sub-band for a frame; discrete cosine transforming at least one vector, the at least one vector comprising the at least one spread and/or surround coherence value for a sub-band for the frame; and encoding a first number of components of the discrete cosine transformed vector based on the determined codebook.

An example method of operation includes identifying a loudspeaker array profile defining characteristics of a loudspeaker array stored in memory, identifying a three-dimensional venue geometry value stored in the memory, defining virtual receivers to simulate acoustic characteristics within the venue geometry, defining a number of passive acoustic filter permutations to perform within a range of passive acoustic filter settings, and each passive acoustic filter setting is unique and has one or more passive acoustic filters to apply to one or more loudspeakers in the loudspeaker array, selecting performance criteria to apply to the loudspeaker array to represent its sound coverage uniformity at a given location throughout the venue geometry, calculating the performance criteria of the loudspeaker array via a passive acoustic filter setting selected from one or more of the passive acoustic filter permutations by performing a simulation with the passive acoustic filter settings, identifying an optimized passive acoustic filter setting from a specific permutation, with which the loudspeaker array achieves optimal uniform sound coverage in the venue geometry, and applying the optimized passive acoustic filter setting to the loudspeaker array.

An apparatus including circuitry configured for: defining at least one ambience audio representation, the ambience audio representation includes at least one respective diffuse background audio signal and at least one parameter, the at least one parameter associated with the at least one respective diffuse background audio signal and further associated with at least one frequency range or at least one part of the frequency range, at least one time period or at least one part of the time period and a directional range for a defined position within an audio field, wherein the at least one component representation is configured to be used in rendering an ambiance audio signal by a 6- degrees-of-freedom or enhanced 3-degrees-of-freedom Tenderer by processing, based on the at least one ambience audio representation and a listener position and/or direction, the respective diffuse background audio signal.

A method to encode audio signals is provided for use with an audio capture device that includes multiple microphones having a spatial arrangement on the device, a method to encode audio signals comprising: receiving multiple microphone signals corresponding to the multiple microphones; determining a number and directions of arrival of directional audio sources represented in the one or more microphone signals; determining one of an active microphone signal component and a passive microphone signal component, based upon the determined number and directions of arrival; determining the other of the active microphone signal component and the passive microphone signal component, based upon the determined one of the active input spatial audio signal component and the passive input spatial audio signal component; encoding the active microphone signal component; encoding the passive microphone signal component.

The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

A stereo signal processing method and apparatus, where the method includes performing delay estimation on a stereo signal of a current frame to determine an inter-channel time difference of the current frame, identifying a sign of the inter-channel time difference of the current frame is different from a sign of an inter-channel time difference of a previous frame of the current frame, performing delay alignment processing on the first-channel signal of the current frame based on the inter-channel time difference of the current frame, and performing delay alignment processing on the second-channel signal of the current frame based on the inter-channel time difference of the previous frame.

An apparatus includes a receiver and an up-mixer. The receiver is configured to receive a bitstream that includes an encoded mid signal and encoded stereo parameter information. The encoded stereo parameter information represents a first value of a stereo parameter and a second value of the stereo parameter. The first value is associated with a first frequency range. The second value is associated with a second frequency range that is distinct from the first frequency range. The up-mixer is configured to perform an up-mix operation on a frequency-domain decoded mid signal generated from the encoded mid signal. A particular value based on the first value and the second value is applied to the frequency-domain decoded mid signal during the up-mix operation.