Abstract: A method for processing an audio signal, comprising: receiving the audio signal; and processing the received audio signal, wherein the audio signal is processed according to a scheme comprising: comparing a size information of at least two blocks of A+1 level with a size information of a block of A level corresponding to the at least two of A+1 level; and, determining the at least two blocks of A+1 level as an optimum block if the size information of the at least two blocks of A+1 level is less than the size information of the block of A level is disclosed.

Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19 or 20 unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19 or 20 of unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: In one embodiment, the method includes receiving an audio signal including a prediction residual of a block of digital audio data and coded coefficient values. Table index information is obtained from the digital audio data. The table index information identifies a table from a plurality of tables to select. A set of prediction coefficient values are reconstructed from the coded-coefficient values. This reconstruction includes selecting a table including offset values and entropy parameters from the plurality of tables based on the table index information, wherein 16, 19, or 20 unique offsets are associated with coded-coefficients of the first twenty coefficients, first entropy decoding the coded-coefficient values using entropy codes defined by the entropy parameters from the selected table, and calculating a set of prediction coefficient values based on the offset values from the selected table and the decoded coded-coefficient values.

Abstract: In one embodiment, the method includes receiving the audio signal having a plurality of random access units. The random access unit includes one or more frames and at least one of the frames is a random access frame. The random access frame is a frame encoded such that previous frames are not necessary to decode the random access frame. The embodiment further includes reading location information from the audio signal. The location information indicates whether random access unit size information is stored or not in the audio signal. If the random access unit size information is stored, the location information further indicates a location where the random access unit size information is stored in the audio signal. Random access unit size information is read according to the location information. The random access unit size information indicates a distance between random access frames in bytes. The random access units are decoded based on the random access size information.

Abstract: In one embodiment, the method includes reading random access information from the audio signal. The random access information indicates whether or not random access operation is allowed in the audio signal. If the random access operation is allowed, the audio signal has a plurality of random access units. A random access unit includes one or more of frames and at least one of the frames is a random access frame. The random access frame is a frame encoded such that previous frames are not necessary to decode the random access frame, and the random access information further indicates a distance between random access frames in frames. The random access frame is decoded based on the random access information.

Abstract: In one embodiment, the method includes receiving an audio signal having configuration information and at least one channel, and reading an indicator from the configuration information. The indicator indicates whether or not channel mapping information is included in the configuration information. The channel mapping information is read from the configuration information if the indicator indicates that the channel mapping information is included in the configuration information. The channel mapping information indicates to which speaker in a reproduction device to map each channel in the audio signal. The channel is decoded based on the channel mapping information.

Abstract: In one embodiment, the method includes receiving an audio data frame having at least first and second channels. The first and second channels have been independently subdivided into blocks if the first and second channels are not correlated with each other, and the first and second channels have been synchronously subdivided into blocks if the first and second channels are correlated with each other and difference coding is used. The embodiment further includes obtaining subdivision information from the audio data frame. The subdivision information includes first information and second information. The first information indicates whether the first and second channels are independently subdivided or synchronously subdivided, and the second information indicates how the subdividing is performed. The first and second channels are decoded based on the obtained subdivision information.

Abstract: In one embodiment of the method, a block of digital audio data is stored in a buffer, and a set of parcor values are calculated for the block of digital audio data. Then, the parcor values are entropy encoded for transmission. The entropy coding includes selecting a table including offset values and entropy parameters from a plurality of tables based on a sampling rate of the block of digital audio data, wherein each offset is associated with a coded-coefficient and is different for each coded-coefficient in the first twenty coefficients, calculating parcor residual values based on offset values from the selected table, and encoding the parcor residual values using entropy codes defined by entropy parameters from the selected table.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data are provided jointly if the first and second channel data are paired with each other. The embodiment further includes obtaining block information indicating the block type, and lossless decoding the first and second channel data based on the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data include a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. The method further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. Block information indicating the block type is obtained. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The first and second channel data is provided jointly if the first and second channel data are paired with each other. The method further includes obtaining frame length information indicating a length of the audio frame data, obtaining block information indicating a block type, and lossless decoding the first and second channel data based on the frame length information and the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. Block information indicating the block type is obtained. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data is lossless decoded based on the block information.

Abstract: In one embodiment, the method includes receiving audio frame data having at least first and second channel data. The first and second channel data includes a plurality of blocks, where the blocks are classified by a block type. The embodiment further includes obtaining frame length information indicating a length of the audio frame data, and obtaining block information indicating the block type. The block information corresponds to the first and second channel data being common when the first and second channel data are paired. The first and second channel data are lossless decoded based on the frame length information and the block information.

Abstract: In one embodiment, at least one channel in a frame of the audio signal is subdivided into a plurality of blocks such that at least two of the blocks having different lengths. A length of the frame is a user defined value and is determined within a predetermined value. Furthermore, information indicating the subdivision of the channel into the blocks is generated.

Abstract: In one embodiment, at least one channel in a frame of the audio signal is subdivided into a plurality of blocks such that at least two of the blocks having different lengths, and information indicating the subdivision of the channel into the blocks is generated.

Abstract: In one embodiment, the method includes receiving the audio signal including a block of audio data partitioned into N sub-blocks, and restoring a plurality of code parameters s(0), s(1), . . . , s(N?1), respectively. The restoring step includes detecting s(0) from the audio signal, where s(0) represents the code parameter of the first sub-block; detecting a difference s(i)?s(i?1) from the audio signal for i=1, . . . N?1, where s(i) representing the code parameter of each sub-block following the first sub-block. The difference s(i)?s(i?1) is encoded by using first entropy code. The restoring step further includes calculating s(i) for i=1, . . . , N?1 using s(0) and the detected differences, and the method further includes decoding the N sub-blocks using the restored code parameters.

Abstract: In one embodiment, the method includes receiving an audio data frame having at least one channel. The channel is subdivided into a plurality of blocks, and at least two of the blocks are capable of having different lengths. The embodiment further includes obtaining indicator information indicating whether determining of a prediction order for each block is allowed, and determining the prediction order from the audio signal indicating the prediction order for each block if the indicator information indicating that determining of the prediction order for each block is allowed. The channel is decoded using the prediction order.

Abstract: In one embodiment, sample information and frame length information are obtained from the audio signal. The sample information indicates a total number of audio data samples for each channel in the audio signal, and the frame length information indicates a number of samples in a frame of each channel. An optimum prediction order is determined for each block based on a maximum permitted prediction order and a length of the block, where a prediction order is the number of linear prediction coefficients. The optimum prediction order is selected as a minimum one of the global prediction order and the local prediction order. The global prediction order is determined based on the maximum permitted prediction order, and the local prediction order is determined based on the length of the block.