摘要:
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.
摘要:
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.
摘要:
A method, medium, and apparatus with scalable channel decoding. The method includes recognizing the configuration of channels or speakers, calculating the respective number of same path 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 respective number of decoding levels.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.
摘要:
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.