Abstract: A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame.

Abstract: A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame.

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: 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 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: 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: 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 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: 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: 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 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: A method and an apparatus for recovering a line spectrum pair (LSP) parameter of a spectrum region when frame loss occurs during speech decoding and a speech decoding apparatus adopting the same are provided. The method of recovering an LSP parameter in speech decoding includes: if it is determined that a received speech packet has an erased frame, converting an LSP parameter of a previous good frame (PGF) of the erased frame or LSP parameters of the PGF and a next good frame (NGF) of the erased frame into a spectrum region and obtaining a spectrum envelope of the PGF or spectrum envelopes of the PGF and NGF; recovering a spectrum envelope of the erased frame using the spectrum envelope of the PGF or the spectrum envelopes of the PGF and NGF; and converting the recovered spectrum envelope of the erased frame into an LSP parameter of the erased frame.

Abstract: A voice encoding/decoding method and apparatus. A voice encoder includes: a quantization selection unit generating a quantization selection signal; and a quantization unit extracting a linear prediction coding (LPC) coefficient from an input signal, converting the extracted LPC coefficient into a line spectral frequency (LSF), quantizing the LSF with a first LSF quantization unit or a second LSF quantization unit based on the quantization selection signal, and converting the quantized LSF into a quantized LPC coefficient. The the quantization selection signal selects the first LSF quantization unit or second LSF quantization unit based on characteristics of a synthesized voice signal in previous frames of the input signal.

Abstract: A method, medium, and apparatus for generating a record sentence to establish a speech corpus, including generating a synthesized sentence of speech and synthesis information related to speech synthesis by performing speech synthesis for a predetermined sentence of text, selecting an unseen sentence including an unseen unit according to the synthesis information, generating a weight indicating a recording priority of the unseen unit included in the selected unseen sentence, and generating a record sentence by combining the unseen unit with the speech synthesis information according to the generated weight.