Abstract:
A bitstream reconstruction method and apparatus are provided. The method includes constructing an initial bitstream by rendering sound source information and geometry information within a reference radius from an initial location of a user accessing a virtual space into spatial audio, collecting location information according to a movement of the user within the virtual space, and reconstructing, based on a relationship between the reference radius and a movement radius identified according to the collected location information, the initial bitstream constructed by corresponding to the initial location of the user.
Abstract:
A rendering method of an object-based audio signal and an apparatus for performing the same are provided. The rendering method of an object-based audio signal includes obtaining a rendered audio signal, performing clipping prevention on the rendered audio signal using a first limiter, mixing a signal output by the first limiter using a mixer, and performing clipping prevention on the mixed signal using a second limiter.
Abstract:
A method of rendering object-based audio and an electronic device performing the method are provided. The method includes identifying a bitstream, determining a reference distance of an object sound source based on the bitstream, determining a minimum distance for applying distance-dependent attenuation, based on the reference distance, and determining a gain of object-based audio included in the bitstream based on the reference distance and the minimum distance.
Abstract:
The present invention relates to a method and an apparatus for binaural rendering an audio signal using variable order filtering in frequency domain. To this end, provided are a method for processing an audio signal including: receiving an input audio signal; receiving a set of truncated subband filter coefficients for filtering each subband signal of the input audio signal, the set of truncated subband filter coefficients being constituted by one or more FFT filter coefficients generated by performing FFT by a predetermined block size; generating at least one subframe for each subband; generating at least one filtered subframe for each subband; performing inverse FFT on the filtered subframe for each subband; and generating a filtered subband signal by overlap-adding the transformed subframe for each subband and an apparatus for processing an audio signal using the same.
Abstract:
Provided is an encoding apparatus for integrally encoding and decoding a speech signal and a audio signal, and may include: an input signal analyzer to analyze a characteristic of an input signal; a stereo encoder to down mix the input signal to a mono signal when the input signal is a stereo signal, and to extract stereo sound image information; a frequency band expander to expand a frequency band of the input signal; a sampling rate converter to convert a sampling rate; a speech signal encoder to encode the input signal using a speech encoding module when the input signal is a speech characteristics signal; a audio signal encoder to encode the input signal using a audio encoding module when the input signal is a audio characteristic signal; and a bitstream generator to generate a bitstream.
Abstract:
A Unified Speech and Audio Codec (USAC) that may process a window sequence based on mode switching is provided. The USAC may perform encoding or decoding by overlapping between frames based on a folding point when mode switching occurs. The USAC may process different window sequences for each situation to perform encoding or decoding, and thereby may improve a coding efficiency.
Abstract:
An audio rendering method and an electronic device performing the same are disclosed. The disclosed audio rendering method includes determining an air absorption attenuation amount of an audio signal based on a recording distance included in metadata of the audio signal and a source distance between a sound source of the audio signal and a listener; and rendering the audio signal based on the air absorption attenuation amount.
Abstract:
Disclosed is a method and apparatus for processing an audio signal based on an extent sound source. The method includes identifying information on a reference area of the extent sound source and information on a position of a listener, determining a position of a virtual sound source within the extent sound source based on a relationship between the position of the listener and the reference area of the extent sound source, and rendering an audio signal based on the determined position of the virtual sound source, wherein the reference area may be determined based on a position and a size of the extent sound source.
Abstract:
An encoding apparatus and a decoding apparatus in a transform between a Modified Discrete Cosine Transform (MDCT)-based coder and a different coder are provided. The encoding apparatus may encode additional information to restore an input signal encoded according to the MDCT-based coding scheme, when switching occurs between the MDCT-based coder and the different coder. Accordingly, an unnecessary bitstream may be prevented from being generated, and minimum additional information may be encoded.
Abstract:
The present invention relates to a system for transmitting and receiving audio, particularly, to a method and apparatus for transmitting and receiving of object-based audio contents, which packetizes audio objects having the same characteristic.To achieve the above, the present invention includes filtering a plurality of ESs according to common information, adding a packet header to the respective filtered ESs and generate ES packets, aggregating all the generated ES packets and then adding a multi-object packet header to the aggregated ES packets to generate an object packet, and multiplexing the generated object packet, packetizing the multiplexed object packet according to a transmitting media and transmitting the packetized object packet.