H04S2420/03

Parametric audio decoding

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.

Audio encoding apparatus and method, audio decoding apparatus and method, and audio reproducing apparatus

An audio encoding apparatus and method that encodes hybrid contents including an object sound, a background sound, and metadata, and an audio decoding apparatus and method that decodes the encoded hybrid contents are provided. The audio encoding apparatus may include a mixing unit to generate an intermediate channel signal by mixing a background sound and an object sound, a matrix information encoding unit to encode matrix information used for the mixing, an audio encoding unit to encode the intermediate channel signal, and a metadata encoding unit to encode metadata including control information of the object sound.

Method for outputting audio signal using scene orientation information in an audio decoder, and apparatus for outputting audio signal using the same
11310616 · 2022-04-19 · ·

A method for decoding a bitstream by an apparatus, includes obtaining a decoded audio signal and metadata from the bitstream, the metadata comprising scene orientation information; and rendering the decoded audio signal based on the scene orientation information, wherein the scene orientation information is information for a direction of a video scene related to the decoded audio signal.

Methods and systems for rendering object based audio

Methods for generating an object based audio program, renderable in a personalizable manner, and including a bed of speaker channels renderable in the absence of selection of other program content (e.g., to provide a default full range audio experience). Other embodiments include steps of delivering, decoding, and/or rendering such a program. Rendering of content of the bed, or of a selected mix of other content of the program, may provide an immersive experience. The program may include multiple object channels (e.g., object channels indicative of user-selectable and user-configurable objects), the bed of speaker channels, and other speaker channels. Another aspect is an audio processing unit (e.g., encoder or decoder) configured to perform, or which includes a buffer memory which stores at least one frame (or other segment) of an object based audio program (or bitstream thereof) generated in accordance with, any embodiment of the method.

Parametric reconstruction of audio signals

An encoding system encodes an N-channel audio signal (X), wherein N≥3, as a single-channel downmix signal (Y) together with dry and wet upmix parameters ({tilde over (C)}, {tilde over (P)}). In a decoding system, a decorrelating section outputs, based on the downmix signal, an (N−1)-channel decorrelated signal (Z); a dry upmix section maps the downmix signal linearly in accordance with dry upmix coefficients (C) determined based on the dry upmix parameters; a wet upmix section populates an intermediate matrix based on the wet upmix parameters and knowing that the intermediate matrix belongs to a predefined matrix class, obtains wet upmix coefficients (P) by multiplying the intermediate matrix by a predefined matrix, and maps the decorrelated signal linearly in accordance with the wet upmix coefficients; and a combining section combines outputs from the upmix sections to obtain a reconstructed signal ({circumflex over (X)}) corresponding to the signal to be reconstructed.

Methods and systems for rendering audio based on priority

Embodiments are directed to a method of rendering adaptive audio by receiving input audio comprising channel-based audio, audio objects, and dynamic objects, wherein the dynamic objects are classified as sets of low-priority dynamic objects and high-priority dynamic objects, rendering the channel-based audio, the audio objects, and the low-priority dynamic objects in a first rendering processor of an audio processing system, and rendering the high-priority dynamic objects in a second rendering processor of the audio processing system. The rendered audio is then subject to virtualization and post-processing steps for playback through soundbars and other similar limited height capable speakers.

Stereo signal processing method and apparatus

A stereo signal processing 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.

Representing spatial audio by means of an audio signal and associated metadata

There is provided encoding and decoding methods for representing spatial audio that is a combination of directional sound and diffuse sound. An exemplary encoding method includes inter alia creating a single- or multi-channel downmix audio signal by downmixing input audio signals from a plurality of microphones in an audio capture unit capturing the spatial audio; determining first metadata parameters associated with the downmix audio signal, wherein the first metadata parameters are indicative of one or more of: a relative time delay value, a gain value, and a phase value associated with each input audio signal; and combining the created downmix audio signal and the first metadata parameters into a representation of the spatial audio.

Methods and systems for rendering audio based on priority

Embodiments are directed to a method of rendering adaptive audio by receiving input audio comprising channel-based audio, audio objects, and dynamic objects, wherein the dynamic objects are classified as sets of low-priority dynamic objects and high-priority dynamic objects, rendering the channel-based audio, the audio objects, and the low-priority dynamic objects in a first rendering processor of an audio processing system, and rendering the high-priority dynamic objects in a second rendering processor of the audio processing system. The rendered audio is then subject to virtualization and post-processing steps for playback through soundbars and other similar limited height capable speakers.

METHOD FOR GENERATING FILTER FOR AUDIO SIGNAL, AND PARAMETERIZATION DEVICE FOR SAME
20210368286 · 2021-11-25 ·

The present invention relates to a method for generating a filter for an audio signal and a parameterization device for the same, and more particularly, to a method for generating a filter for an audio signal, to implement filtering of an input audio signal with a low computational complexity, and a parameterization device therefor.

To this end, provided are a method for generating a filter for an audio signal, including: receiving at least one binaural room impulse response (BRIR) filter coefficients for binaural filtering of an input audio signal; converting the BRIR filter coefficients into a plurality of subband filter coefficients; obtaining average reverberation time information of a corresponding subband by using reverberation time information extracted from the subband filter coefficients; obtaining at least one coefficient for curve fitting of the obtained average reverberation time information; obtaining flag information indicating whether the length of the BRIR filter coefficients in a time domain is more than a predetermined value; obtaining filter order information for determining a truncation length of the subband filter coefficients, the filter order information being obtained by using the average reverberation time information or the at least one coefficient according to the obtained flag information and the filter order information of at least one subband being different from filter order information of another subband; and truncating the subband filter coefficient by using the obtained filter order information and a parameterization device therefor.