Abstract: A voice signal processing method includes acquiring a near-end noise signal and a near-end voice signal by using at least one microphone, acquiring a far-end voice signal according to an incoming call, determining a noise control parameter and a voice signal change parameter based on at least one of information about the near-end voice signal, information about the near-end noise signal, or information about the far-end voice signal, generating an anti-phase signal of the near-end noise signal based on the noise control parameter, changing the far-end voice signal to improve articulation of the far-end voice signal based on information related to at least one of the voice signal change parameter, the near-end noise signal, or the anti-phase signal, and outputting the anti-phase signal and the changed far-end voice signal.

Abstract: A voice signal processing method according to an embodiment of the present disclosure for overcoming the problem includes: acquiring a real-time near-end noise signal; acquiring a far-end voice signal according to an incoming call; measuring subjective speech quality and perceptual-objective speech quality of test signals generated based on a reference signal and the real-time near-end noise signal; selecting at least one speech quality enhancement method based on the subjective speech quality and the perceptual-objective speech quality, and determining parameters that are to be applied to the selected at least one speech quality enhancement method; and enhancing speech quality of the far-end voice signal by using the selected at least one speech quality enhancement method, based on the determined parameters, :herein the test signals are generated by mixing the acquired real-time near-end noise signal with the reference signal whose speech quality is enhanced by applying a combination of parameter values to s

Abstract: Proposed is a method and apparatus for determining a weighting function for quantizing a linear predictive coding (LPC) coefficient and having a low complexity. The weighting function determination apparatus may convert an LPC coefficient of a mid-subframe of an input signal to one of a immittance spectral frequency (ISF) coefficient and a line spectral frequency (LSF) coefficient, and may determine a weighting function associated with an importance of the ISF coefficient or the LSF coefficient based on the converted ISF coefficient or LSF coefficient.

Abstract: A frame error concealment method is provided that includes predicting a parameter by performing a regression analysis on a group basis for a plurality of groups formed from a first plurality of bands forming an error frame and concealing an error in the error frame by using the parameter predicted on a group basis.

Abstract: A method and apparatus for performing coding and decoding for high-frequency bandwidth extension. The coding apparatus may classify a coding mode of a low-frequency signal of an input signal based on characteristics of the low-frequency signal of an input signal, perform code excited linear prediction coding or audio coding on the LPC excitation signal of the low-frequency signal of an input signal, and perform time-domain (TD) extension coding or frequency-domain (FD) extension coding on a high-frequency signal of an input signal. When the FD extension coding is performed, the coding apparatus may generate a base excitation signal for a high band using an input spectrum, obtain an energy control factor of a sub-band in a frame using the base excitation signal and the input spectrum, generate an energy signal based on the input spectrum and the energy control factor, for the sub-band in the frame, and quantize the energy signal.

Abstract: Disclosed are a user terminal device and a control method therefor. The control method for a user terminal device according to the present invention comprises the steps of: determining state information of a satellite device connected with a user terminal device when a call request message is received from another terminal device; determining communication configuration information with the another terminal device on the basis of the state information of the satellite device, and transmitting a call response message including the communication configuration information to the another terminal device; and receiving audio data from the another terminal device according to a communication scheme corresponding to the communication configuration information. Therefore, in comparison with the prior art, the present invention can reduce a call negotiation time between user terminal devices, and thus reduce a call cost.

Abstract: An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.

Abstract: A quantization apparatus comprises: a first quantization module for performing quantization without an inter-frame prediction; and a second quantization module for performing quantization with an inter-frame prediction, and the first quantization module comprises: a first quantization part for quantizing an input signal; and a third quantization part for quantizing a first quantization error signal, and the second quantization module comprises: a second quantization part for quantizing a prediction error; and a fourth quantization part for quantizing a second quantization error signal, and the first quantization part and the second quantization part comprise a trellis structured vector quantizer.

Abstract: A quantization device includes: a trellis-structured vector quantizer which quantizes a first error vector between an N-dimensional (here, “N” is two or more) subvector and a first predictive vector; and an inter-frame predictor which generates a first predictive vector from the quantized N-dimensional subvector, wherein the inter-frame predictor uses a predictive coefficient comprising an N×N matrix and performs an inter-frame prediction using the quantized N-dimensional subvector of a previous stage.

Abstract: The present invention relates to a method and an apparatus for encoding and decoding spectrum coefficients in the frequency domain. The spectrum encoding method may comprise the steps of: selecting an encoding type on the basis of bit allocation information of respective bands; performing zero encoding with respect to a zero band; and encoding information of selected significant frequency components with respect to respective non-zero bands. The spectrum encoding method enables encoding and decoding of spectrum coefficients which is adaptive to various bit-rates and various sub-band sizes. In addition, a spectrum can be encoded using a TCQ method at a fixed bit rate using a bit-rate control module in a codec that supports multiple rates. Encoding performance of the codec can be maximised by encoding high performance TCQ at a precise target bit rate.

Abstract: Provided is a method for processing a signal adaptively to a noisy environment, the method including: determining a class related to enhancement of intelligibility, based on a noise spectrum of a reception side or a near-end device and a received voice spectrum; generating guide information for the enhancement of intelligibility based on the noise spectrum, the voice spectrum, and the determined class; and generating a changed voice spectrum by applying the guide information to the voice spectrum.

Abstract: A frame error concealment method is provided that includes predicting a parameter by performing a regression analysis on a group basis for a plurality of groups formed from a first plurality of bands forming an error frame and concealing an error in the error frame by using the parameter predicted on a group basis.

Abstract: Provided are a method and an apparatus for increasing call quality while reducing consumption of bit rates in immersive multimedia communication. A method of transmitting, by a terminal device, an audio signal to a counterpart terminal device, includes: extracting head tracking information of a user of the counterpart terminal device from a packet received from the counterpart terminal device; predicting a head direction of the user of the counterpart terminal device after a predetermined time period, based on the extracted head tracking information; generating audio output signals by combining audio input signals received via a plurality of microphones, based on the predicted head direction; and transmitting information about the predicted head direction and a sound frame in which the audio output signals are compressed, to the counterpart terminal device.

Abstract: A method and apparatus for performing coding and decoding for high-frequency bandwidth extension. The coding apparatus may classify a coding mode of a low-frequency signal of an input signal based on characteristics of the low-frequency signal of an input signal, perform code excited linear prediction coding or audio coding on the LPC excitation signal of the low-frequency signal of an input signal, and perform time-domain (TD) extension coding or frequency-domain (FD) extension coding on a high-frequency signal of an input signal. When the FD extension coding is performed, the coding apparatus may generate a base excitation signal for a high band using an input spectrum, obtain an energy control factor of a sub-band in a frame using the base excitation signal and the input spectrum, generate an energy signal based on the input spectrum and the energy control factor, for the sub-band in the frame, and quantize the energy signal.

Abstract: Provided are a terminal device and method of performing a call function transmitting ambient audio with high sensitivity. A terminal device performing a call function with at least one external device via a network may include a receiver configured to receive at least one of an audio transmission signal and a video transmission signal to be transmitted to the external device; a processor configured to analyze at least one of the audio transmission signal and the video transmission signal, select one of a speech mode and an audio mode, based on a result of the analysis, and compress the audio transmission signal, based on the selected mode; a communicator configured to transmit the compressed audio transmission signal to the external device, and receive an audio reception signal from the external device; and an output unit configured to output the audio reception signal.

Abstract: An apparatus and method for encoding and decoding a signal for high frequency bandwidth extension are provided. An encoding apparatus may down-sample a time domain input signal, may core-encode the down-sampled time domain input signal, may transform the core-encoded time domain input signal to a frequency domain input signal, and may perform bandwidth extension encoding using a basic signal of the frequency domain input signal.

Abstract: An apparatus determining a weighting function for line prediction coding coefficients quantization converts a linear prediction coding (LPC) coefficient of an input signal into one of a line spectral frequency (LSF) coefficient and an immitance spectral frequency (ISF) coefficient and determines a weighting function associated with one of an importance of the ISF coefficient and importance of the LSF coefficient using one of the converted ISF coefficient and the converted LSF coefficient.

Abstract: The present invention relates to a method and an apparatus for encoding and decoding spectrum coefficients in the frequency domain. The spectrum encoding method may comprise the steps of: selecting an encoding type on the basis of bit allocation information of respective bands; performing zero encoding with respect to a zero band; and encoding information of selected significant frequency components with respect to respective non-zero bands. The spectrum encoding method enables encoding and decoding of spectrum coefficients which is adaptive to various bit-rates and various sub-band sizes. In addition, a spectrum can be encoded using a TCQ method at a fixed bit rate using a bit-rate control module in a codec that supports multiple rates. Encoding performance of the codec can be maximised by encoding high performance TCQ at a precise target bit rate.

Abstract: A spectrum encoding method includes selecting an important spectral component in band units for a normalized spectrum and encoding information of the selected important spectral component for a band, based on a number, a position, a magnitude and a sign thereof. A spectrum decoding method includes obtaining from a bitstream, information about an important spectral component for a band of an encoded spectrum and decoding the obtained information of the important spectral component, based on a number, a position, a magnitude and a sign of the important spectral component.

Abstract: A method of generating a wideband signal may include estimating a high-band spectrum parameter from a reconstructed narrowband signal based on a combination of at least two mapping schemes; estimating a high-band excitation signal based on the reconstructed narrowband signal; generating a high-band signal based on the estimated high-band spectrum parameter and the estimated high-band excitation signal; and generating a wideband signal by synthesizing the reconstructed narrowband signal with the high-band signal.