Abstract: A method and device to arithmetically decode scalable bit sliced arithmetic coding (BSAC) audio data are provided. The arithmetic decoding method includes checking whether all side information of a last layer of the BSAC audio data is to be decoded, and performing BSAC smart decoding and terminating the decoding when all the side information of the last layer is not decoded, and terminating decoding of the last layer when all the side information of the last layer is decoded. The BSAC smart decoding includes checking whether a symbol to be decoded is determined regardless of data read after the truncated bitstream, continuing the decoding when decoding can be performed regardless of data read after the truncated bitstream, and determining that ambiguity occurs and terminating the decoding, when the symbol to be decoded is determined dependent on the data read after the truncated bitstream. Accordingly, byte level scalability can be provided in a MPEG-4 BSAC decoder.

Abstract: A scalable audio data arithmetic decoding method, medium, and apparatus, and a method, medium, and apparatus truncating an audio data bitstream. The arithmetic decoding method of decoding a scalable arithmetic coded symbol may include arithmetic decoding of a symbol by using the symbol and a probability value for the symbol desired to be decoded, and determining whether or not to continue decoding by checking an ambiguity indicating whether or not decoding of the symbol to be decoded is completed. According to a method, medium, and apparatus of the present invention, data to which scalability is applied when arithmetic coding is performed in MPEG-4 scalable lossless audio coding can be efficiently decoded. Even when a bitstream is truncated, a decoding termination point can be known such that additional decoding of the truncated part can be performed.

Abstract: An encoder and decoder to encode one or more input signals into a scalable codec and to decode the scalable codec, and encoding and decoding methods using a bitstream with a layered structure in the scalable codec change a top coding bit rate to encode the input signals according to a network status, and the bitstream is decoded by analyzing the top coding bit rate included in the bitstream.

Abstract: A method and an apparatus to encode and decode a speech signal using a code excited linear prediction (CELP) algorithm. In order to reduce a bit rate without degrading performance in an enhancement layer based on CELP, each of a fixed codebook of a core layer and a fixed codebook of the enhancement layer is divided into a plurality of spaces. The spaces of the fixed codebook of the enhancement layer excludes a space corresponding to a least distorted space determined from among the spaces of the fixed codebook of the core layer are searched.

Abstract: A method, medium, and apparatus with scalable channel decoding. The method includes recognizing the configuration of channels or speakers, calculating the number of decoding levels for each multi-channel signal using the recognized configuration of the channels or speakers, and performing decoding and up-mixing according to the calculated number of decoding levels.

Abstract: A method and apparatus to quantize and dequantize an input signal, and a method and apparatus to encode and decode an input signal. The method of quantizing an input signal includes determining a quantization scale type according to a distribution feature of the input signal, and quantizing the input signal according to the determined quantization scale type. Accordingly, when a number of assigned bits is small in an encoding process, signal distortion can be minimized without an increase in complexity or having to use large sized additional information in order to determine an optimum scale. In addition, the input signal can be encoded by considering a trade-off of a distortion rate corresponding to the number of assigned bits of the input signal.

Abstract: A method, medium, and system generating a 3-dimensional (3D) stereo signal in a decoder by using a surround data stream. According to such a method, medium, and system, a head related transfer function (HRTF) is applied in a quadrature mirror filter (QMF) domain, thereby generating a 3D stereo signal by using a surround data stream.

Abstract: Surround audio decoding for selectively generating an audio signal from a multi-channel signal. In the surround audio decoding, a down-mixed signal, e.g., as down-mixed by an encoding terminal, is selectively up-mixed to a stereo signal or a multi-channel signal, by generating spatial information for generating the stereo signal, using spatial information for up-mixing the down-mixed signal to the multi-channel signal.

Abstract: A method, medium, and system decoding and/or encoding multiple channels. Accordingly, down-mixed multiple channels can be decoded/up-mixed to a left channel and a right channel during a first stage, thereby enabling a high quality sound output even in scalable channel decoding.

Abstract: An system, method, and method of encoding/decoding a multi-channel audio signal, including a decoding level generation unit producing decoding-level information that helps a bitstream including a number of audio channel signals and space information to be decoded into a number of audio channel signals, wherein the space information includes information about magnitude differences and/or similarities between channels, and an audio decoder decoding the bitstream according to the decoding-level information. Accordingly, even a single input bitstream can be decoded into a suitable number of channels depending on the type of a speaker configuration used. Scalable channel decoding can be achieved by partially decoding an input bitstream. In the scalable channel decoding, a decoder may set decoding levels and outputs audio channel signals according to the decoding levels, thereby reducing decoding complexity.

Abstract: A method and device to arithmetically decode scalable bit sliced arithmetic coding (BSAC) audio data are provided. The arithmetic decoding method includes checking whether all side information of a last layer of the BSAC audio data is to be decoded, and performing BSAC smart decoding and terminating the decoding when all the side information of the last layer is not decoded, and terminating decoding of the last layer when all the side information of the last layer is decoded. The BSAC smart decoding includes checking whether a symbol to be decoded is determined regardless of data read after the truncated bitstream, continuing the decoding when decoding can be performed regardless of data read after the truncated bitstream, and determining that ambiguity occurs and terminating the decoding, when the symbol to be decoded is determined dependent on the data read after the truncated bitstream. Accordingly, byte level scalability can be provided in a MPEG-4 BSAC decoder.

Abstract: Adaptive time/frequency-based audio encoding and decoding apparatuses and methods. The encoding apparatus includes a transformation & mode determination unit to divide an input audio signal into a plurality of frequency-domain signals and to select a time-based encoding mode or a frequency-based encoding mode for each respective frequency-domain signal, an encoding unit to encode each frequency-domain signal in the respective encoding mode, and a bitstream output unit to output encoded data, division information, and encoding mode information for each respective frequency-domain signal. In the apparatuses and methods, acoustic characteristics and a voicing model are simultaneously applied to a frame, which is an audio compression processing unit. As a result, a compression method effective for both music and voice can be produced, and the compression method can be used for mobile terminals that require audio compression at a low bit rate.

Abstract: A method, medium, and apparatus hierarchically coding/decoding audio data, such as bit sliced arithmetic coding (BSAC), such that payloads of audio data and extension data can be grouped and interleaved according to priority so that some groups of the payloads are dropped, and the remainder of groups are transmitted Therefore, extension data that is more important than a top layer of audio data, in terms of reproducing of original sounds, can be transmitted with priority.

Abstract: A method, medium, and apparatus encoding/decoding audio data in which audio data is hierarchically encoded, and at least one extension data of the audio data is encoded using at least one encoding method, and decoding is performed in the same manner, thereby ensuring fine grain scalability (FGS) and unlimited extendibility of the audio data.

Abstract: A method and apparatus to encode/decode a low bit-rate audio signal. The method of encoding a low bit-rate audio signal, includes quantizing and losslessly-encoding a specific frequency component of an audio signal in a frequency domain, generating codebooks using the audio signal in the frequency domain, detecting an envelope of a frequency component of the audio signal other than the specific frequency component in a specific band unit and quantizing and losslessly-encoding the detected envelope of the other frequency component, selecting a codebook that is most similar to the other frequency component of the audio signal to be encoded from among the generated codebooks and determining a codebook index (fine structure), losslessly-encoding the determined codebook index, and generating a bit stream using the specific frequency component, the envelope of the other frequency component, and the determined codebook index.

Abstract: An method and apparatus to extract an audio signal having an important spectral component (ISC) and a low bit-rate audio signal coding/decoding method using the method and apparatus to extract the ISC. The method of extracting the ISC includes calculating perceptual importance including an SMR (signal-to-mark ratio) value of transformed spectral audio signals by using a psychoacoustic model, selecting spectral signals having a masking threshold value smaller than that of the spectral audio signals using the SMR value as first ISCs, and extracting a spectral peak from the audio signals selected as the ISCs according to a predetermined weighting factor to select second ISCs. Accordingly, the perceptual important spectral components can be efficiently coded so as to obtain high sound quality at a low bit-rate.

Abstract: A scalable audio data arithmetic decoding method, medium, and apparatus, and a method, medium, and apparatus truncating an audio data bitstream. The arithmetic decoding method of decoding a scalable arithmetic coded symbol may include arithmetic decoding of a symbol by using the symbol and a probability value for the symbol desired to be decoded, and determining whether or not to continue decoding by checking an ambiguity indicating whether or not decoding of the symbol to be decoded is completed. According to a method, medium, and apparatus of the present invention, data to which scalability is applied when arithmetic coding is performed in MPEG-4 scalable lossless audio coding can be efficiently decoded. Even when a bitstream is truncated, a decoding termination point can be known such that additional decoding of the truncated part can be performed.

Abstract: An apparatus and method of low bit rate encoding and reproducing. The method includes transforming input audio signals in a time domain into spectral signals in a frequency domain, extracting important-spectrum components from the spectral signals in the frequency domain, and quantizing the important-spectrum components, extracting residual-spectrum components other than the important-spectrum components from the spectral signals in the frequency domain, and calculating and quantizing a noise level of the residual-spectrum components, and encoding the quantized important-spectrum components and the quantized noise level losslessly, and outputting encoded bitstreams.

Abstract: A method and apparatus for transforming an audio signal, a method and apparatus for adaptively encoding an audio signal, a method and apparatus for inversely transforming an audio signal, and a method and apparatus for adaptively decoding an audio signal. The method of transforming an audio signal includes determining a transform unit into which the audio signal in a time domain is to be transformed into an audio signal in a frequency domain, and transforming the audio signal into an audio signal in the frequency domain according to the determined transform units using a window coefficient other than 0. Accordingly, it is possible to minimize distortion of the audio signal when encoding the audio signal even at a high bit rate while increasing efficiency of compression.

Abstract: An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal.