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:
Disclosed is a binaural rendering method and apparatus for decoding a multichannel audio signal. The binaural rendering method may include: extracting an early reflection component and a late reverberation component from a binaural filter; generating a stereo audio signal by performing binaural rendering of a multichannel audio signal base on the early reflection component; and applying the late reverberation component to the generated stereo audio signal.
Abstract:
Disclosed are a device and method for audio signal processing. The audio signal processing device according to an embodiment includes a receiver configured to receive a bitstream corresponding to a compressed audio signal and a processor. The processor may be configured to generate a real restoration signal or a complex restoration signal by performing inverse quantization on real data of the bitstream or complex data of the bitstream, generate a result of real Frequency Domain Noise Shaping (FDNS) synthesis or a result of complex FDNS synthesis by performing FDNS synthesis on the real restoration signal or the complex restoration signal, and generate a restored audio signal by performing frequency-to-time transform on the result of the real FDNS synthesis or the result of the complex FDNS synthesis.
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:
Provided is an encoding apparatus including a memory configured to store instructions and a processor electrically connected to the memory and configured to execute the instructions, wherein the processor may be configured to perform a plurality of operations, when the instructions are executed by the processor, wherein the plurality of operations may include obtaining an input audio signal, generating an embedded audio signal by embedding signal components of a second frequency band of the input audio signal in a first frequency band of the input audio signal, generating additional information associated with the first frequency band and the second frequency band, generating an encoded audio signal by encoding the embedded audio signal, and formatting the encoded audio signal and the additional information into a bitstream.
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 method of rendering object-based audio and an electronic device for performing the method are disclosed. The method includes identifying metadata of the object-based audio, determining whether the metadata includes a parameter set for an atmospheric absorption effect for each distance, and rendering the object-based audio, using a distance between the object-based audio and a listener obtained using the metadata and the atmospheric absorption effect according to an effect of a medium attenuation based on the parameter, when the metadata includes the parameter.
Abstract:
A generative adversarial network-based audio signal generation model for generating a high quality audio signal may comprise: a generator generating an audio signal with an external input; a harmonic-percussive separation model separating the generated audio signal into a harmonic component signal and a percussive component signal; and at least one discriminator evaluating whether each of the harmonic component signal and the percussive component signal is real or fake.
Abstract:
Provided are a method of training a neural network model, a method of recognizing an acoustic event and an acoustic direction, and an electronic device for performing the methods. A method of training a neural network model according to an example embodiment includes generating a heatmap indicating an acoustic event and an acoustic direction in which the acoustic event occurs by using training data, outputting a result of recognizing the acoustic event and the acoustic direction by inputting a feature extracted using the training data into a neural network model for recognizing the acoustic event and the acoustic direction of the training data, and training the neural network model by using the result and the heatmap.
Abstract:
Provided are a method of recognizing sound, a method of training a sound recognition model, and an electronic device performing the same methods. A method of training a sound recognition model according to an example embodiment may include converting training data labeled with a sound class into a feature vector, storing the feature vector in a feature queue, transferring the feature vector stored in the feature queue to a block queue according to an operation of a feature vector transfer timer, inputting the feature vector of the block queue into a sound recognition model trained to predict the sound class and storing an output result in a result queue, transferring the feature vector stored in the feature queue corresponding to timing at which the result is output to the block queue by the feature vector transfer timer when the result is output.