Abstract: The present invention discloses an encoding apparatus using a Discrete Cosine Transform (DCT) scanning, which includes: a mode selection means for selecting an optimal mode for intra prediction; an intra prediction means for performing intra prediction onto video inputted based on the mode selected in the mode selection means; a DCT and quantization means for performing DCT and quantization onto residual coefficients of a block outputted from the intra prediction means; and an entropy encoding means for performing entropy encoding onto DCT coefficients acquired from the DCT and quantization by using a scanning mode decided based on pixel similarity of the residual coefficients.

Abstract: Provided is an apparatus and method for coding and decoding multi-object audio signals with various channels and providing backward compatibility with a conventional spatial audio coding (SAC) bitstream. The apparatus includes: an audio object coding unit for coding audio-object signals inputted to the coding apparatus based on a spatial cue and creating rendering information for the coded audio-object signals, where the rendering information provides a coding apparatus including spatial cue information for audio-object signals; channel information of the audio-object signals; and identification information of the audio-object signals, and used in coding and decoding of the audio signals.

Abstract: Provided are an apparatus and method for coding and decoding a multi-object audio signal. The apparatus includes a down-mixer for down-mixing the audio signals into one down-mixed audio signal and extracting supplementary information including header information and spatial cue information for each of the audio signals, a coder for coding the down-mixed audio signal, and a supplementary information coder for generating the supplementary information as a bit stream. The header information includes identification information for each of the audio signals and channel information for the audio signals.

Abstract: The present research relates to controlling rendering of multi-object or multi-channel audio signals. The present research provides a method and apparatus for controlling rendering of multi-object or multi-channel audio signals based on spatial cues in a process of decoding the multi-object or multi-channel audio signals. To achieve the purpose, the method suggested in the research controls rendering in a spatial cue domain in the process of decoding the multi-object or multi-channel audio signals.

Abstract: An apparatus and method for deciding an adaptive noise level for bandwidth extension are provided. The apparatus includes a noise level decider for deciding a high-band noise level for bandwidth extension according to tonality of an input signal, a pitch frequency analyzer for detecting a pitch frequency of the input signal and analyzing correlation between the detected pitch frequency and a frequency channel, and a noise level controller for adaptively controlling the decided high-band noise level based on the analyzed correlation of the pitch frequency and the frequency channel.

Abstract: A Unified Speech and Audio Codec (USAC) for adjusting an overlap area of a window based on a transition is provided. To increase an encoding efficiency, encoding may be performed by overlapping relatively long windows. Additionally, when a transition is generated between frames, an overlap area of a window may be reduced based on the transition, thereby preventing a noise from occurring due to the transition.

Abstract: Provided is a method for controlling synchronizing downmix signals and MPEG surround side information signals by controlling a delay according to the kind of downmix audio signals in an MPEG surround decoder. When multi-channel audio signals are restored using an HE-AAC decoder and a low-power MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. Anther delay unit delays spatial parameters to compensate for a delay caused in QMF and Nyquist banks when time-domain downmix signals are used. Also, when multi-channel audio signals are restored using an HE-AAC decoder and a high-quality MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter.

Abstract: Disclosed are an apparatus and a method for separating sound sources capable of learning distributions of corresponding sound sources based on the assumption that specific sound sources have specific distributions based on interchannel correlation parameter in audio signals providing space perception through a plurality of channels to separate an amount corresponding to energy contribution of the corresponding sound sources from mixture signals. Exemplary embodiments of the present invention can more precisely predict the channel distributions of the specific sound sources included in the input mixture signals and more accurately separate sound sources than a method for separating a sound source based on the channel according to the related art, under conditions that general channel distribution information of the specific sound sources are approximately modeled.

Abstract: A known information compression apparatus and method for reducing a size of known information without missing information required to separate a sound source are provided. The known information compression apparatus may include a segment dividing unit to divide known information including sound source information of each musical instrument into a plurality of segments, and a compressed information generating unit to downmix the segments and to generate compressed information.

Abstract: Disclosed are a method and a system of separating and extracting unified major sound sources from a mixed musical signal. A unified sound source separation system includes a first sound source separation unit to separate a first sound source having unique time-domain and frequency-domain characteristics from a mixed musical signal which includes a plurality of sound sources using time-domain and frequency-domain characteristics, and a second sound source separation unit to separate a second sound source existing in a predetermined stereo sound image position from the mixed musical signal using stereo channel information.

Abstract: Provided are an apparatus and method of separating, from a mixed signal, a sound source generated using a rhythm musical instrument based on characteristics of the rhythm musical instrument repeated in an aspect of time. The apparatus may include a separation unit to separate a plurality of mixed signals into a plurality of segments, a Nonnegative Matrix Partial Co-Factorization (NMPCF) analysis unit to perform an NMPCF analysis on the plurality of segments, and to obtain a plurality of entity matrices based on the analysis result, a target instrument signal separating unit to separate, from the mixed signals, a target instrument signal, by calculating an inner product between the plurality of entity matrices, and a signal association unit to associate the target instrument signals separated from each of the plurality of segments.

Abstract: Disclosed is an LPC residual signal encoding/decoding apparatus of an MDCT based unified voice and audio encoding device. The LPC residual signal encoding apparatus analyzes a property of an input signal, selects an encoding method of an LPC filtered signal, and encode the LPC residual signal based on one of a real filterbank, a complex filterbank, and an algebraic code excited linear prediction (ACELP).

Abstract: An encoding apparatus and a decoding apparatus in a transform between a Modified Discrete Cosine Transform (MDCT)-based coder and a hetero 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 hetero coder. Accordingly, an unnecessary bitstream may be prevented from being generated, and minimum additional information may be encoded.

Abstract: A module capable of appropriately selecting a linear predictive coding (LPC)-based or a code excitation linear prediction (CELP)-based speech or audio encoder and a transform-based audio encoder according to a feature of an input signal is a module that performs as a bridge for overcoming a performance barrier between a conventional LPC-based encoder and an audio encoder. Also, an integral audio encoder that provides consistent audio quality regardless of a type of the input audio signal can be designed based on the module.

Abstract: Provided is an apparatus for integrally encoding and decoding a speech signal and an audio signal. An encoding apparatus for integrally encoding a speech signal and an audio signal, may include: a module selection unit to analyze a characteristic of an input signal and to select a first encoding module for encoding a first frame of the input signal; a speech encoding unit to encode the input signal according to a selection of the module selection unit and to generate a speech bitstream; an audio encoding unit to encode the input signal according to the selection of the module selection unit and to generate an audio bitstream; and a bitstream generation unit to generate an output bitstream from the speech encoding unit or the audio encoding unit according to the selection of the module selection unit.

Abstract: Provided are an apparatus and method for providing an object based audio file, and an apparatus and method for playing back an object based audio file. The object based audio file producing apparatus may include a bitstream generator to generate a bitstream about an object based audio file including a plurality of audio object frames and a file header for an object based audio service; and a bitstream transmitter to transmit the bitstream to the object based audio file playback apparatus. The plurality of audio object frames may include a frame storing a audio source in which all of a plurality of audio frames is mixed and a frame storing each of the audio objects.

Abstract: Provided are an apparatus and method of separating, from a mixed signal, a sound source generated using a rhythm musical instrument based on characteristics of the rhythm musical instrument repeated in an aspect of time. The apparatus may include a separation unit to separate a plurality of mixed signals into a plurality of segments, a Nonnegative Matrix Partial Co-Factorization (NMPCF) analysis unit to perform an NMPCF analysis on the plurality of segments, and to obtain a plurality of entity matrices based on the analysis result, a target instrument signal separating unit to to separate, from the mixed signals, a target instrument signal, by calculating an inner product between the plurality of entity matrices, and a signal association unit to associate the target instrument signals separated from each of the plurality of segments.

Abstract: Provided are a method and apparatus for structuring a bitstream for an object-based audio service, and an apparatus for encoding the bitstream. A method of structuring a bitstream, may include: configuring the bitstream by separating the bitstream into a file header and frames of audio objects that are separated using a sound source separation scheme; and storing, in the file header, reproduction level information of audio objects.

Abstract: Provided is a method for controlling synchronizing downmix signals and MPEG surround side information signals by controlling a delay according to the kind of downmix audio signals in an MPEG surround decoder. When multi-channel audio signals are restored using an HE-AAC decoder and a low-power MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter. Anther delay unit delays spatial parameters to compensate for a delay caused in QMF and Nyquist banks when time-domain downmix signals are used. Also, when multi-channel audio signals are restored using an HE-AAC decoder and a high-quality MPEG surround decoder and complex QMF signals outputted from the HE-AAC decoder are used as downmix signals, a delay unit compensates for a delay caused in a real-to-complex converter.

Abstract: Provided is a method and apparatus for generating a side information bitstream of a multi-object audio signal. The apparatus for generating a side information bitstream of a multi-object audio signal includes a spatial cue information input unit configured to receive spatial cue information generated in an encoder of the multi-object audio signal, a preset information input unit configured to receive preset information for the multi-object audio signal, and a side information bitstream generator configured to generate the side information bitstream based on the spatial cue information and the preset information. The side information bitstream includes a header region and a frame region, and the preset information is included in the frame region.