Classifications

• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/008—Multichannel audio signal coding or decoding using interchannel correlation to reduce redundancy, e.g. joint-stereo, intensity-coding or matrixing
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/0017—Lossless audio signal coding; Perfect reconstruction of coded audio signal by transmission of coding error
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/0212—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using orthogonal transformation
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/022—Blocking, i.e. grouping of samples in time; Choice of analysis windows; Overlap factoring
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
  • G10L19/12—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a code excitation, e.g. in code excited linear prediction [CELP] vocoders
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/16—Vocoder architecture
  • G10L19/167—Audio streaming, i.e. formatting and decoding of an encoded audio signal representation into a data stream for transmission or storage purposes
• • G—PHYSICS
  • G11—INFORMATION STORAGE
  • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
  • G11B27/00—Editing; Indexing; Addressing; Timing or synchronising; Monitoring; Measuring tape travel
  • G11B27/10—Indexing; Addressing; Timing or synchronising; Measuring tape travel
  • G11B27/102—Programmed access in sequence to addressed parts of tracks of operating record carriers
  • G11B27/105—Programmed access in sequence to addressed parts of tracks of operating record carriers of operating discs
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S1/00—Two-channel systems
  • H04S1/007—Two-channel systems in which the audio signals are in digital form
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/005—Correction of errors induced by the transmission channel, if related to the coding algorithm
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
  • G10L19/032—Quantisation or dequantisation of spectral components
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/06—Determination or coding of the spectral characteristics, e.g. of the short-term prediction coefficients
• • G—PHYSICS
  • G10—MUSICAL INSTRUMENTS; ACOUSTICS
  • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
  • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
  • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
  • G10L19/16—Vocoder architecture
  • G10L19/18—Vocoders using multiple modes
  • G10L19/24—Variable rate codecs, e.g. for generating different qualities using a scalable representation such as hierarchical encoding or layered encoding
• • G—PHYSICS
  • G11—INFORMATION STORAGE
  • G11B—INFORMATION STORAGE BASED ON RELATIVE MOVEMENT BETWEEN RECORD CARRIER AND TRANSDUCER
  • G11B20/00—Signal processing not specific to the method of recording or reproducing; Circuits therefor
  • G11B20/10—Digital recording or reproducing
  • G11B20/10527—Audio or video recording; Data buffering arrangements
  • G11B2020/10537—Audio or video recording
  • G11B2020/10546—Audio or video recording specifically adapted for audio data

Definitions

• the present invention relates to a method for processing audio signal, and more particularly to a method and apparatus of
• MPEG MP3 or AAC
• DVD audio and Super CD Audio include proprietary
• Lossless audio coding permits the compression of digital audio data without any loss in quality- due to a perfect reconstruction of the original signal.
• the present invention relates a method of processing an audio signal.
• a channel in a frame of the audio signal is subdivided into a plurality of blocks, and at least two of the blocks have different lengths.
• An optimum prediction order for each block is determined based on a permitted prediction order and a length of the block.
• the determination of the optimum prediction order includes determining a global prediction order based on the permitted prediction order, determining a local prediction order based on the length of the block, and selecting a minimum one of the global prediction order and the local prediction order as the optimum prediction order.
• the global prediction order may be determined as equal to,
• the local prediction order may be determined as equal to,
• the channel is subdivided according to a subdivision hierarchy.
• each level is associated with a different block length. For example, a
• a channel has a length of N, then the
• channel is subdivided into the plurality of blocks such that
• each block has a length of one of N/2, N/4, N/8, N/16 and N/32.
• information is generated such that a length
• the information depends on a number of levels in the subdivision hierarchy.
• the information may be
• each information bit may be associated with a level in the
• subdivision hierarchy and may be associated with a block at
• each information bit indicates
• the method further includes predicting
• the prediction is performed by
• this progressive prediction process may be any progressive prediction process.
• this progressive prediction process may be any progressive prediction process.
• the present invention further relates to methods and apparatuses for encoding an audio signal, and to methods and
• FIG. 1 is an example illustration of an encoder according to
• FIG. 2 is an example illustration of a decoder according to an embodiment of the present invention.
• FIG. 3 is an example illustration of a bitstream structure of
• FIG. 4 is an example illustration of a conceptual view of a
• FIG. 5 is an example illustration of a block switching
• FIG. 6 is an example illustration of block switching methods
• FIG. 1 is an example illustration of an encoder 1 according to the present invention.
• a partitioning part 100 partitions the input audio data into
• each channel may be further
• a buffer 110 stores block and/or frame samples partitioned by the partitioning part 100.
• a coefficient estimating part 120 estimates an optimum set of
• the order of the predictor can be adaptively chosen as
• the coefficient estimating part 120 calculates a set of
• a quantizing part 130 quantizes the set of
• a first entropy coding part 140 calculates parcor residual
• the entropy parameters are chosen from an optimal table.
• optimal table is selected from a plurality of tables based on
• the plurality of tables are predefined for a plurality of sampling
• a coefficient converting part 150 converts the quantized
• a predictor 160 estimates current prediction values from the
• the buffer 110 and a prediction value estimated in the predictor 160.
• a second entropy coding part 180 codes the prediction residual
• the second entropy coding part 180 may code the
• a multiplexing part 190 multiplexes coded prediction
• the encoder 1 also provides a cyclic redundancy check (CRC) checksum, which is supplied mainly for the decoder to verify
• the decoded data On the encoder side, the CRC can be used to calculate the decoded data.
• the CRC On the encoder side, the CRC can be used to calculate the decoded data.
• the CRC On the encoder side, the CRC can be used to calculate the decoded data.
• Additional encoding options include flexible block switching
• the encoder 1 1
• the joint channel coding is used to
• FIG. 2 is an example illustration of a decoder 2 according to
• FIG. 2 shows the lossless audio signal decoder which is significantly less
• a demultiplexing part 200 receives an audio signal and
• part 210 decodes the parcor residual values using entropy
• a coefficient converting part 230 converts
• predictor 240 estimates a prediction residual of the block of
• An adder 250 is provided to convert digital audio data using the LPC coefficients.
• An assembling part 260 assembles the decoded
• the decoder 2 decodes the coded prediction residual
• FIG. 3 is an example illustration of a bitstream structure
• a compressed audio signal including a plurality of channels
• the bitstream consists of at least one audio frame including a
• Each channel is sub ⁇
• Each sub-divided block has a different size
• the coding data within a subdivided block For example, the coding data within a subdivided block
• partition is identical for both channels, and blocks are
• bitstream configuration syntax (Table 6) indicates whether
• joint stereo (channel difference) is on or off, and a
• the block partition for each channel is independent .
• An aspect of the present invention relates to subdividing each channel into a plurality of blocks prior to using the actual
• block switching method referred to as a "block switching method” .
• FIG. 4 is an example illustration of a conceptual view of a
• FIG. 4 illustrates a method of
• each channel is provided in a single frame, each
• channel may be subdivided (or partitioned) to up to 32 blocks,
• the partitioning part 100 shown in FIG. 1 is performed by the partitioning part 100 shown in FIG. 1. Furthermore, as described above, the
• Each channel of N samples is either encoded using one full length
• N B N
• N B N
• each channel of a frame may be
• FIG. 4 illustrates a channel which can be
• N/32 may be possible within a channel according to the
• each block results from a subdivision of a superordinate block of double length.
• partition into N/4 + N/2 + N/4 may not be possible (e.g., block switching examples shown in FIGs. 5 (e) and 5 described
• each block has a length equal to one of ,
• N is the length of the channel
• m is an integer greater
• p represents a number of the levels in the subdivision hierarchy.
• bitstream includes information indicating block switching
• settings are made so that a minimum block size
• level 0 block switching which is referred to as a level 0 block switching.
• the first block switching information For example, the first
• block switching information may be represented by a 2-bit
• the second block switching information may be
• bit_info represented by a "bs_info” field which is expressed by any one of 8 bits, 16 bits, and 32 bits within the syntax shown in
• the total number of bits being allocated for the second block switching information is decided based upon the level value of
• Table 1 Block switching levels.
• mapping each bit within the second block switching
• the bs_info field may include up to 4 bytes in accordance with
• the first bit may be reserved for indicating independent or synchronous block switching, which is described in more detail below in the Independent/Synchronous Block
• FIGs. 5(a)-5(f) illustrate different block
• N B N/8, and the bs_info consists of one byte.
• bs_info are set if a block is further subdivided. For example,
• first block of length N/2 is further split ((0)110%) into two
• FIG. 5(f) could not have been obtained by subdividing a block
• FIGs. 6 (a) - 6 (c) are example illustrations of block switching
• FIG. 6 (a) illustrates an example where block switching has not been performed for channels 1, 2, and
• FIG. 6 (b) illustrates an example in which two channels
• channels 1 and 2 configure one channel pair, and block
• channel pair refers to two arbitrary audio channels.
• the decision on which channels are grouped into channel pairs can be
• channel may be identical for all channels, the block switching
• the channels may be divided into blocks
• channels of a channel pair may be block switched synchronously.
• the channels are block switched (i.e., divided into blocks) in the same manner.
• FIG. 1 In synchronous block switching, the channels are block switched (i.e., divided into blocks) in the same manner.
• the blocks may be interleaved. If the two channels of a
• the described method of independent or synchronous block switching may be applied to a multi-channel group having
• a number of channels equal to or more than 3 channels .
• channels of a multi-channel group are not correlated with each
• each channel of the multi-channel group may be switched independently.
• the "bs_info" field is used as the information for
• a particular bit within the "bs__info" field may be used.
• the first bit of the w bs_info" field is set to "1" .
• bit of the "bs_info” field is set as "0" .
• FIGs. 6 (a) , 6Cb), and 6(c) will now be described
• channels 1 and 2 configure a channel
• both channels 1 and 2 are split into
• the interleaving may be beneficial (or
• a block of one channel (e.g.,
• channels 1 and 2 configure a channel
• channel 1 is split into
• channel data may be arranged separately.
• Joint channel coding also called joint stereo, can be used to
• the channels can be rearranged by
• the encoder in order to assign suitable channel pairs.
• lossless audio codec also supports a more complex scheme for exploiting inter-channel
• the present invention relates to audio lossless coding and is able to supports random access. Random access stands for fast
• the encoder needs to insert a frame that
• Random access frame is referred to as a "random access frame" .
• no samples from previous frames may be used for prediction.
• a ⁇ random_access field is used as information for indicating whether random access is allowed
• the 8-bit "random_access” field designates the number of frames configuring a random access
• Random_access > 0 random access is supported.
• a 32-bit u ra_unit_size" field is included in the bitstream and
• configuration syntax (Table 6) may further include information
• configuration syntax (Table 6) may also be referred to as
• the "ra_flag" field may also be used to indicate the "ra_flag" field.
• an audio signal includes
• each random access unit containing one or more audio data frames, one of which is a random access frame, wherein
• the configuration information includes first general
• the random access unit size information indicating a distance between two adjacent random access unit
• method of decoding an audio signal includes receiving the
• each random access unit containing one
• an audio signal includes multi-channels information according to the present invention.
• each channel may be mapped at a one-to-one correspondence with
• the "chan_config_info” field includes
• channels field is equal to or more than "2", this indicates that the channel corresponds to one of multi-channels .
• present invention also includes information indicating whether
• Table 2 Channel configuration.
• an audio signal includes multiple or multi-channels according to the present invention. Therefore, when performing encoding, information on the number of multichannels configuring one frame and information on the number of samples for each channel are inserted in the bitstream and transmitted.
• a 32-bit "samples" field is used as information indicating the total number of audio data samples configuring each channel.
• a 16-bit w frame_length" field is used as information indicating the number of samples for each channel within the corresponding frame.
• a 16-bit value of the "frame_length" field is determined by a value used by the encoder, and is referred to as a user-defined value.
• the user-defined value is arbitrarily determined upon the encoding process.
• the frame number of each channel should first be obtained. This value is obtained according to the algorithm shown below.
• samples field is an exact multiple of the number of samples
• the multiple value becomes the total number of frames. However, if the total number of samples decided by the
• samples field is not an exact multiple of the number of
• the encoder may freely decide and
• samples For each channel and the number of samples (frame_length” field)
• the decoder may
• the predictor 160 shown in FIG. 1 The predictor 160 shown in FIG. 1
• the second entropy coding part 180 performs entropy
• predictor coefficient values are entropy coded by the first
• Linear prediction is used in many applications for speech and
• FIR FIR Response
• the current sample of a time-discrete signal x ⁇ can be
• K is the order of the predictor.
• the coefficients should be transmitted.
• forward adaptation in this case, the coefficients should be transmitted.
• the backward adaptation procedure has
• Another aspect of forward-adaptive prediction is to determine
• bit rate R e for the residual.
• bit rate R e for the residual.
• the total bit rate can be
• the prediction order K is also determined by the coefficient estimating part 120.
• bitstream the bitstream and then transmitted.
• the configuration syntax (Table 6) includes information
• the "max_order" field becomes the final order applied to all of the blocks .
• the optimum order (opt_order) is decided based upon the value
• the opt_order for each block may be decided considering the size of the
• the opt_order value being
• the present invention relates to higher
• short block length e.g. 4096 S- 1024 or 8192 & 2048
• this factor can be increased (e.g., up to 32), enabling a
• Table 8 can also be up to 10 bits. The actual number of bits in a particular block may depend on the maximum order
• prediction order may be smaller than a global prediction order.
• the local prediction order is determined from
• prediction order is determined from the ⁇ max_order" K max in the
• the "opt_order” field is determined on 8 bits (instead
• the opt_order may be determined based on
• opt_order min (global prediction order, local prediction order) ;
• a channel are predicted.
• a first sample of a current block is predicted using the last K samples of a previous block.
• the K value is determined from the opt_order which is derived from the above-described equation.
• the current block is a first block of the channel, no samples from the previous block are used.
• the above-described progressive order type of prediction is very advantageous when used in the random access frame. Since the random access frame corresponds to a reference frame of the random access unit, the random access frame does not perform prediction by using the previous frame sample. Namely, this progressive prediction technique may be applied at the beginning of the random access frame.
• predictor coefficients h k is not very efficient for
• coefficient estimating part 120 As described above, for example, the coefficient estimating part 120 is processed
• the first two parcor coefficients ( ⁇ 1 and ' 2 correspondingly) are quantized by using the following functions:
• Entropy Coding As shown in FIG. 1, two types of entropy coding are applied in the present invention. More specifically, the first entropy coding part 140 is used for coding the above-described predictor coefficients. And, the second entropy coding part 180 is used for coding the above-described audio original samples and audio residual samples.
• the two types of entropy coding will now be described in detail.
• the related art Rice code is used as the first entropy coding method according to the present invention. For example,
• the first entropy coding part 140 which, in turn, are encoded by using the first entropy coding part 140, e.g., the Rice code method.
• the corresponding offsets and parameters of Rice code used in this process can be globally chosen from one of the sets shown in Table 3, 4 and 5 below.
• a table index i.e., a 2-bit "coef_table”
• the first entropy decoding part 220 reconstructs the
• ⁇ k ⁇ k +o ⁇ sQt k
• rQ is an empirically determined mapping table (not shown).
• mapping table may vary with implementation
• the first entropy coding are provided according to the sampling frequency.
• the sampling frequency may be divided to 48kHz, 96kHz, and 192kHz.
• table can also be chosen by other criteria.
• Table 3 Rice code parameters used for encoding of
• Table 4 Rice code parameters used for encoding of
• Table 5 Rice code parameters used for encoding of
• the present invention contains two different modes of the coding method applied to the second entropy coding part 180 of
• FIG. 1 which will now be described in detail.
• the indices of the applied codes are transmitted, as
• the encoder can use a more complex and
• the encoding of residuals is accomplished by splitting the distribution in two categories .
• the two types include
• tails are simply re-centered (i.e., for e(n) > e max ,
• the BGMC first splits the
• the BGMC encodes MSBs using block Gilbert-Moore (arithmetic) codes.
• the BGMC transmits LSBs using direct fixed-lengths
• transmitted LSBs may be selected such that they only slightly
• the configuration syntax (Table 6) first includes a 1-bit
• ⁇ bgmc_mode 0
• the "sb_part" field corresponds to information related
• the "ec_sub” field indicates the number of sub-
• ⁇ bgmc_mode + sbjpart 1" signifies that the Rice code or the BGMC code is used to partition the block to sub-
• Second entropy coding part 180 are coded by second entropy coding part 180 using a difference coding method.
• An example of using the Rice code will now be
• configuration syntax may form a header periodically placed in
• bitstream may form a header of each frame; etc.
• Table 8 shows a block-data syntax
• Table 7 Frame data syntax.
• Table 8 Block data syntax.
• the lossless audio codec is compared with
• high-definition material i.e., 96 kHz / 24-bit and above.
• the audio signal encoder of the present invention is the audio signal encoder of the present invention
• Pentium-M depending on audio format (kHz/bits) and ALS
• the codec is designed to offer a large range of complexity
• audio data can be decoded even on hardware with very low computing power.
• the decoder may be

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• 2006
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