Abstract: A digital broadcasting system which is robust against an error when mobile service data is transmitted and a method of processing data are disclosed. The mobile service data is subjected to an additional coding process and the coded mobile service data is transmitted. Accordingly, it is possible to cope with a serious channel variation while applying robustness to the mobile service data.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: A signal processing apparatus and method thereof are disclosed. The present invention includes receiving a low frequency downmix signal including a multi channel signal, phase shift information and spatial information corresponding to parameter band of the low frequency downmix signal, generating the multi channel signal by applying the spatial information based on the parameter band to a whole frequency downmix signal, generating estimated phase shift information of a parameter band by using the phase shift information, and generating a phase shift multi channel signal by shifting a phase of the multi channel signal based on the phase shift information and the estimated phase shift information. Accordingly, it is able to efficiently reproduce a phase or delay difference, which is difficult to be efficiently reproduced by a decorrelator, in a manner of shifting a phase of a decoded audio or speech signal based on phase shift information.

Abstract: A digital broadcasting system which is robust against an error when mobile service data is transmitted and a method of processing data are disclosed. The mobile service data is subjected to an additional coding process and the coded mobile service data is transmitted. Accordingly, it is possible to cope with a serious channel variation while applying robustness to the mobile service data.

Abstract: The present invention relates to a method and apparatus for processing a signal. An object of the present invention devised to solve the problem lies on a method and apparatus for processing a signal, which allows a signal having optimized signal transmission efficiency to be transmitted/received.

Abstract: A digital broadcasting system which is robust against an error when mobile service data is transmitted and a method of processing data are disclosed. The mobile service data is subjected to an additional coding process and the coded mobile service data is transmitted. Accordingly, it is possible to cope with a serious channel variation while applying robustness to the mobile service data.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a downmix processing information using the object information and the mix information; processing the downmix signal using the downmix processing information; and, generating a multi-channel information using the object information and the mix information, wherein the number of channel of the downmix signal is equal to the number of channel of the processed downmix signal is disclosed.

Abstract: A digital broadcasting system which is robust against an error when mobile service data is transmitted and a method of processing data are disclosed. The mobile service data is subjected to an additional coding process and the coded mobile service data is transmitted. Accordingly, it is possible to cope with a serious channel variation while applying robustness to the mobile service data.

Abstract: An apparatus for processing an encoded signal and method thereof are disclosed, by which an audio signal can be compressed and reconstructed in higher efficiency. An audio signal processing method includes the steps of identifying whether a type of an audio signal is a music using first type information, if the type of the audio signal is not the music signal, identifying whether the type of the audio signal is a speech signal or a mixed signal using second type information, and if the type of the audio signal is determined as either the speech signal or the mixed signal, reconstructing the audio signal according to a coding scheme applied per frame using coding identification information. If the type of the audio signal is the music signal, the first type information is received only. If the type of the audio signal is the speech signal or the mixed signal, both of the first type information and the second type information are received.

Abstract: A method for processing an audio signal, comprising the steps of extracting an ancillary signal for generating the audio signal and an extension signal included in the ancillary signal from a received bit stream, reading length information of the extension signal from an extension area of the ancillary signal, and generating the audio signal using the ancillary signal. Accordingly, in case of processing the audio signal by the present invention, it is able to reduce a corresponding load of operation to enable efficient processing and enhance a sound quality.

Abstract: A method for processing an audio signal, comprising the steps of extracting an ancillary signal for generating the audio signal and an extension signal included in the ancillary signal from a received bit stream, checking a level of the bit stream, selectively decoding the extension signal according to the level of the bit stream, and generating the audio signal using the ancillary signal. Accordingly, in case of processing the audio signal by the present invention, it is able to reduce a corresponding load of operation to enable efficient processing and enhance a sound quality.

Abstract: A method for processing an audio signal, comprising: receiving a downmix signal, an object information, and a mix information; generating a downmix processing information using the object information and the mix information; processing the downmix signal using the downmix processing information; and, generating a multi-channel information using the object information and the mix information, wherein the number of channel of the downmix signal is equal to the number of channel of the processed downmix signal is disclosed.

Abstract: A method for processing an audio signal, comprising the steps of extracting an ancillary signal for generating the audio signal, an extension signal included in the ancillary signal, and header identification information indicating whether a header is included in the ancillary signal from a received bit stream, reading length information of the extension signal from the header if the header is included in the ancillary signal according to the header identification information, skipping decoding of the extension signal or not using a result of the decoding based on the length information, and generating the audio signal using the ancillary signal. Accordingly, in case of processing the audio signal by the present invention, it is able to reduce a corresponding load of operation to enable efficient processing and enhance a sound quality.

Abstract: The present invention relates to a method and apparatus for processing a signal. An object of the present invention devised to solve the problem lies on a method and apparatus for processing a signal, which allows a signal having optimized signal transmission efficiency to be transmitted/received.

Abstract: An audio signal processing method is disclosed. The audio signal processing method includes receiving a residual and long term prediction information, performing inverse frequency mapping with respect to the residual to generate a synthesized residual, and performing long term synthesis based on the synthesized residual and the long term prediction information to generate a synthesized audio signal of a current frame, wherein the long term prediction information comprises a final prediction gain and a final pitch lag, the final pitch lag has a range starting with 0, and the long term synthesis is performed based on a synthesized audio signal of a frame comprising a preceding frame.

Abstract: A method for processing a media signal, comprising: receiving, by an audio processing apparatus, an audio signal including a first channel signal and a second channel signal; estimating center sound by applying a band-pass filter to the first channel signal and the second channel signal; obtaining a first ambient sound by subtracting the center sound from the first channel signal; obtaining a second ambient sound by subtracting the center sound from the second channel signal; applying at least one of delay and reverberation filter to at least one of the first ambient sound and the second ambient sound to generate a processed ambient sound; and, generating pseudo surround signal using the center sound and the processed ambient sound is provided.

Abstract: Spatial information associated with an audio signal is encoded into a bitstream, which can be transmitted to a decoder or recorded to a storage media. The bitstream can include different syntax related to time, frequency and spatial domains. In some embodiments, the bitstream includes one or more data structures (e.g., frames) that contain ordered sets of slots for which parameters can be applied. The data structures can be fixed or variable. The data structure can include position information that can be used by a decoder to identify the correct slot for which a given parameter set is applied. The slot position information can be encoded with either a fixed number of bits or a variable number of bits based on the data structure type.

Abstract: Methods and apparatuses for encoding and decoding a multi-channel audio signal are provided. In the encoding method, spatial information is calculated based on a multi-channel audio signal and a down-mix signal, and a compensation parameter that compensates for the down-mix signal is calculated based on the multi-channel audio signal and the down-mix signal. Thereafter, a bitstream is generated by encoding the spatial information, the compensation parameter, and the down-mix signal and combining the results of the encoding. Therefore, it is possible to prevent deterioration of the quality of sound regarding a multi-channel audio signal by compensating for the multi-channel audio signal using a compensation parameter that compensates for a down-mix signal.

Abstract: A method for processing an audio signal is disclosed. The method for processing an audio signal includes frequency-transforming an audio signal to generate a frequency-spectrum, deciding a weighting per band corresponding to energy per band using the frequency spectrum, receiving a masking threshold based on a psychoacoustic model, applying the weighting to the masking threshold to generate a modified masking threshold, and quantizing the audio signal using the modified masking threshold.

Abstract: Encoding and decoding apparatuses and encoding and decoding methods are provided. The decoding method includes extracting a plurality of encoded signals from an input bitstream, determining which of a plurality of decoding methods is to be used to decode each of the encoded signals, decoding the encoded signals using the determined decoding methods, and synthesizing the decoded signals. Accordingly, it is possible to encode signals having different characteristics at an optimum bitrate by classifying the signals into one or more classes according to the characteristics of the signals and encoding each of the signals using an encoding unit that can best serve the class where a corresponding signal belongs. In addition, it is possible to efficiently encode various signals including audio and speech signals.