JP4441989B2 - Encoding apparatus and encoding method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To judge easily whether or not continuity among several files should be considered when the several files are coded or decoded. SOLUTION: A retrieval condition is set in a step S51 and a variable is initialized in a step S52. When it is determined that there is a file to be retrieved in a step S53, the file is opened in a step S55 and the front part data is read in a step S56. In a step S57 the condition of set threshold value is compared with the data read un. If the data read in is above the set threshold value, warning process is done in a step S58. In a step S59, S60 and S61 final part data is processed. In a step S62 the file is closed and in a step S63 i is incremented, returning to the step S53. Only the file that is given a warning is listened to and thus it is easy to judge whether or not the continuity should be considered.

Description

[0001]
BACKGROUND OF THE INVENTION
  The present invention relates to an encoding apparatus and encoding method relating to a digital signal such as audio data.To the lawRelated.
[0002]
[Prior art]
As a conventional technique related to high-efficiency encoding of audio signals, for example, a time-domain audio signal is blocked per unit time, and a signal on the time axis for each block is converted into a signal on the frequency axis (orthogonal transform). A transform coding method is known which is one of the blocked frequency band division schemes that divide into a plurality of frequency bands and encode each band. In addition, sub-band band coding (sub-band coding) is one of the non-blocking frequency band division methods for coding by dividing the audio signal in the time domain into a plurality of frequency bands without being blocked every unit time. A coding (SBC: Sub Band Coding) method is known.
[0003]
Furthermore, a high-efficiency encoding method that combines the above-described band division encoding and transform encoding is also known. In this method, for example, a signal for each band divided by the band division coding method is orthogonally transformed to a frequency domain signal by the transform coding method, and coding is performed for each band subjected to the orthogonal transformation.
[0004]
Here, examples of the band division filter used in the above-described band division encoding method include a filter such as a QMF (Quadrature Mirror filter). QMF is described in, for example, R.E.Crochiere Digital coding of speech in subbands Bell Syst.Tech. J. Vol.55, No.8 (1976). Also, ICASSP 83, BOSTON Polyphase Quadrature filters-A new subband coding technique Joseph H. Rothweiler describes an equal-bandwidth filter division technique and apparatus such as a polyphase quadrature filter.
[0005]
As orthogonal transform, for example, an input audio signal is blocked in a predetermined unit time (frame), and fast Fourier transform (FFT), cosine transform (DCT), modified DCT transform (MDCT), etc. are performed for each block. Thus, a method for converting the time axis to the frequency axis is known. MDCT is described in, for example, ICASSP 1987 Subband / Transform Coding Using Filter Bank Designs Based on Time Domain Aliasing Cancellation J.P.Princen A.B.Bradley Univ. Of Surrey Royal Melbourne Inst.of Tech.
[0006]
On the other hand, an encoding method using a frequency division width in consideration of human auditory characteristics when quantizing each frequency component obtained by frequency band division is known. That is, a bandwidth called a critical band (critical band) is widely used such that the bandwidth becomes wider as the frequency increases. An audio signal may be divided into a plurality of bands (for example, 25 bands) using such a critical band. According to such a band division method, when data for each band is encoded, encoding by predetermined bit allocation for each band or adaptive bit allocation for each band is performed. For example, when the MDCT coefficient data generated by the MDCT process is encoded by the bit allocation as described above, the adaptive bit is applied to the MDCT coefficient data for each band generated corresponding to each block. Numbers are allocated and encoding is performed under such bit number distributions.
[0007]
As publicly known literature regarding such a bit allocation method and an apparatus for realizing the method, for example, the following can be cited. First, for example, IEEE Transactions of Accoustics, Speech, and Signal Processing, vol.ASSP-25, No.4, August (1977) describes a method of allocating bits based on the signal magnitude for each band. ing. Also, for example, ICASSP 1980 The critical band coder--digital encoding of the perceptual requirements of the auditory system MA Kransner MIT uses auditory masking to obtain the required signal-to-noise ratio for each band and a fixed bit. The method of allocation is described.
[0008]
Also, when encoding for each band, so-called block floating processing is performed to realize more efficient encoding by performing normalization and quantization for each band. For example, when encoding MDCT coefficient data generated by MDCT processing, quantization is performed after performing normalization corresponding to the maximum value of the absolute value of the above-mentioned MDCT coefficient for each band. More efficient encoding is performed. For example, the normalization process is performed as follows. That is, a plurality of types of values numbered in advance are prepared, and among these types of values, those related to normalization for each block are determined by a predetermined calculation process, and the numbers assigned to the determined values Is used as normalization information. Numbering corresponding to a plurality of types of values is performed under a certain relationship, for example, an increase / decrease of the number 1 corresponds to an increase / decrease of 2 dB of the audio level.
[0009]
The highly efficient encoded data generated by the method as described above is decoded as follows. First, processing for generating MDCT coefficient data based on the encoded data is performed with reference to bit allocation information, normalization information, and the like for each band. Time domain data is generated by performing so-called inverse orthogonal transform (IMDCT) based on the MDCT coefficient data. If band division by the band division filter has been performed in the process of high-efficiency encoding, processing for synthesizing time domain data using a band synthesis filter is further performed.
[0010]
In the above-described orthogonal transform MDCT processing used for encoding and inverse orthogonal transform IMDCT processing used for decoding, so-called overlap is used to prevent discontinuity between frames to be processed. Processing is being used. When a certain piece of music is encoded and decoded, matching processing is performed for the start point and end point of the music in consideration of the overlap and the conversion size.
[0011]
The high-efficiency encoding by the above-described method is basically performed in units of music. However, when a large amount of music is subjected to high-efficiency encoding processing, the user is required to perform the next music every time the processing of each music is completed. Since it is inefficient to prompt the start of this process, usually, a process is performed in which a desired music is selected in advance and the automatically selected music is highly efficiently encoded. More specifically, in a distribution server for electronic music distribution, a large amount of PCM files are stored in a hard disk, and high-efficiency encoding processing is performed at high speed by computer software processing.
[0012]
When a high-efficiency encoding process is automatically performed for a large number of music pieces like a distribution server, since high-efficiency encoding is performed in units of music pieces, overlap in orthogonal transformation at the start and end points for each music piece In addition, the adaptation processing considering the conversion size is performed. Depending on the music, when there is a correlation with other music, for example, the start point of the music may maintain continuity with the end point of the other music. As a specific example, in music such as a live version, remix, and dance, there is a case where music pieces are connected without a silent period. Even in such a case, if the adaptation process at the start point and the end point as described above is performed independently for each piece of music, there is a problem that the data after the high-efficiency encoding process loses continuity between pieces of music. Similar problems occur in decoding. When processing what has continuity between music, it is desirable to process inter-music data continuously, without performing the adaptation process in a start point and an end point.
[0013]
[Problems to be solved by the invention]
By the way, the determination as to whether or not to perform the continuous encoding as described above is basically performed by checking the listening of the first and last portions of each piece of music to be actually processed. . However, when it is necessary to process a large number of music pieces, it is necessary to check the listening of all the music pieces, and a great deal of time is consumed. In addition, along with the long-time confirmation work, the judgment ability may be reduced, and the judgment accuracy may be reduced.
[0014]
  Therefore, the object of the present invention is to automatically analyze the data of the first part and the last part of the music selected to be processed and determine whether to consider the continuity associated with encoding or decoding. , Encoding method and encoding method that can be performed easily and accurately and improve work efficiencyThe lawIt is to provide.
[0015]
[Means for Solving the Problems]
  The invention of claim 1 is an encoding device that blocks a plurality of digital audio files for each predetermined length and compresses the block-processed digital audio file,
  Apply compression from multiple digital audio files2 or moreDigital audio filesIn order of compression processingFirst selecting means for selecting;
  Selected by the first selection meansOf the two or more digital audio files, in the order of compression processingA block near the end of a digital audio file positioned in front of the adjacent digital audio file and a block near the start of a digital audio file positioned behind the adjacent digital audio file selected by the first selection means When,AdjacentFirst encoding means for performing an encoding process based on a block straddling two digital audio files;
  Selected by the first selection meansOf the two or more digital audio files, in the order of compression processingBlock near the end of a digital audio file located in front of an adjacent digital audio fileOr a block near the beginning of a digital audio file located behind an adjacent digital audio fileWhen,AdjacentSecond encoding means for performing encoding processing based on blocks straddling two digital audio files;
  Two or more digital audios selected by the first selection meansAn analysis means for analyzing the data at the start and end of the file;
  A second selection unit that selects one of the encoding process in the first encoding unit and the encoding process in the second encoding unit with reference to the analysis result of the analysis unit;MarkEncoding device.
[0017]
  Claim5The present invention is an encoding method in which a plurality of digital audio files are blocked at predetermined lengths and a compression process is performed on the block-processed digital audio file,
  A first selection step of selecting two or more digital audio files to be compressed from a plurality of digital audio files in the order of the compression processing;
  Of the two or more digital audio files selected in the first selection step, a block in the vicinity of the end of the digital audio file located in front of the adjacent digital audio file in the order in which compression processing is performed, and the first selection Encoding processing is performed based on the block near the beginning of the digital audio file located behind the adjacent digital audio file selected in the step and the block straddling the two adjacent digital audio files. A first encoding step;
  Of the two or more digital audio files selected in the first selection step, a block near the end of the digital audio file positioned in front of the adjacent digital audio file in the order in which compression processing is performed or an adjacent digital audio file A second encoding step for performing an encoding process based on a block near the beginning of the digital audio file located behind and a block straddling two adjacent digital audio files;
  An analysis step for analyzing data of start and end points of two or more digital audio files selected in the first selection step;
  A second selection step for selecting one of the encoding process in the first encoding step and the encoding process in the second encoding step with reference to the analysis result of the analysis step;It is an encoding method.
[0020]
According to the invention as described above, it is possible to analyze the continuity of a plurality of files by analyzing the data of the start and end points of the plurality of files to be processed. It is possible to limit the files to be auditioned with reference to the analysis result. Thereby, the processing efficiency can be improved.
[0021]
DETAILED DESCRIPTION OF THE INVENTION
An embodiment of the present invention will be described below with reference to the drawings. In one embodiment, an input digital signal, such as an audio PCM signal, is highly efficient encoded using band division coding (SBC), adaptive transform coding (ATC), and adaptive bit allocation techniques. This high-efficiency encoding technique will be described with reference to FIG.
[0022]
In the high-efficiency encoding device shown in FIG. 1, the input digital signal is divided into a plurality of frequency bands, and orthogonal transform is performed for each frequency band. For each so-called critical bandwidth (critical band) that takes into account human visual characteristics, and in the mid-high range, bit allocation is adaptively assigned to each sub-band of the critical bandwidth considering block floating efficiency. ing. Normally, this block is a quantization noise generation block. Furthermore, in one embodiment, the block size (block length) is adaptively changed according to the input signal before the orthogonal transformation.
[0023]
For example, when the sampling frequency is 44.1 kHz, an audio PCM signal of 0 to 22 kHz is supplied to the band division filter 101 such as a QMF filter via the input terminal 100. The band dividing filter 101 divides the supplied signal into a 0 to 11 kHz band and an 11 kHz to 22 kHz band. The signals in the 11 to 22 kHz band are supplied to an MDCT (Modified Discrete Cosine Transform) circuit 103 and block determination circuits 109, 110, and 111.
[0024]
A signal in the 0 kHz to 11 kHz band is supplied to the band division filter 102. The band division filter 102 divides the supplied signal into a 5.5 kHz to 11 kHz band and a 0 to 5.5 kHz band. The signal in the 5.5 to 11 kHz band is supplied to the MDCT circuit 104 and the block determination circuits 109, 110, and 111. A signal in the 0 to 5.5 kHz band is supplied to the MDCT circuit 105 and the block determination circuits 109, 110, and 111. The band division filters 101 and 102 can be configured using, for example, a QMF filter. The block determination circuit 109 determines the block size based on the supplied signal, and supplies information indicating the determined block size to the MDCT circuit 103 and the output terminal 113.
[0025]
The block determination circuit 110 determines a block size based on the supplied signal, and supplies information indicating the determined block size to the MDCT circuit 104 and the output terminal 115. The block determination circuit 111 determines the block size based on the supplied signal, and sends information indicating the determined block size to the MDCT circuit 105. And supplied to the output terminal 117. The block size block determination circuits 110, 111, and 112 adaptively set the block size (block length) according to the time characteristics and frequency distribution of the supplied signal.
[0026]
The MDCT circuits 103, 104, and 105 perform MDCT processing based on the supplied signal, and generate MDCT coefficient data or spectrum data on the frequency axis. The high-frequency MDCT coefficient data generated by the MDCT circuit 103 or the spectrum data on the frequency axis is subjected to a process for subdividing the critical bandwidth in consideration of the effectiveness of block floating, and then the adaptive bit allocation encoding circuit 106 And supplied to the bit allocation calculation circuit 118. The mid-range MDCT coefficient data generated by the MDCT circuit 104 or the spectrum data on the frequency axis is subjected to processing for subdividing the critical bandwidth in consideration of the effectiveness of block floating, and then the adaptive bit allocation encoding circuit 107. And supplied to the bit allocation calculation circuit 118.
[0027]
The low-frequency MDCT coefficient data generated by the MDCT circuit 105 or the spectrum data on the frequency axis is subjected to a process of grouping for each critical band (critical band), and then applied to the adaptive bit allocation encoding circuit 108 and the bit allocation calculation circuit 118. Supplied. Here, the critical band is a frequency band divided in consideration of human auditory characteristics, and when the pure tone is masked by narrow band noise of the same intensity near the frequency of a certain pure tone, Band noise band. The critical band has the property that the higher the band, the wider the bandwidth. The entire frequency band of 0 to 22 kHz is divided into 25 critical bands, for example.
[0028]
Based on the supplied MDCT coefficient data or spectrum data on the frequency axis, and the block size information, the bit allocation calculation circuit 118 considers the above-described critical band and block floating in consideration of a masking effect and the like as described later. The masking amount, energy, peak value, etc. for each divided band are calculated, and a scale factor indicating the state of block floating and the number of allocated bits are calculated for each band based on the calculation result. The calculated number of allocated bits is supplied to adaptive bit allocation coding circuits 106, 107, and 108. In the following description, each divided band that is a unit of bit allocation is referred to as a unit block.
[0029]
The adaptive bit allocation coding circuit 106 receives from the MDCT circuit 103 according to the block size information supplied from the block determination circuit 109, the number of allocated bits supplied from the bit allocation calculation circuit 118, and the scale factor information as normalization information. The supplied spectrum data or MDCT coefficient data is requantized (normalized and quantized). As a result of such processing, highly efficient encoded data is generated. This high efficiency encoding is supplied to the arithmetic unit 120. The adaptive bit allocation coding circuit 107 receives the spectrum data supplied from the MDCT circuit 104 according to the block size information supplied from the block determination circuit 110, the number of assigned bits supplied from the bit allocation calculation circuit 118, and the scale factor information. Alternatively, a process of requantizing the MDCT coefficient data is performed. As a result of such processing, highly efficient encoded data is generated. This highly efficient encoded data is supplied to the arithmetic unit 121.
[0030]
The adaptive bit allocation encoding circuit 108 uses the block size information supplied from the block determination circuit 110, the number of allocation bits supplied from the bit allocation calculation circuit 118, and the scale factor information to supply spectral data supplied from the MDCT circuit 105. Alternatively, the MDCT coefficient data is requantized. As a result of such processing, highly efficient encoded data is generated. This highly efficient encoded data is supplied to the calculator 122. The normalization information change circuit 119 and the calculators 120, 121, and 122 will be described later.
[0031]
FIG. 2 shows an example of data for each band supplied to the MDCT circuits 103, 104, and 105. By the operation of the block determination circuits 109, 110, and 111, the orthogonal transform block size can be set independently for each band for a total of three data output from the band division filters 101 and 102, and the signal time It is possible to switch the time resolution depending on characteristics, frequency distribution, and the like. That is, when the signal is quasi-stationary in time, Long Mode that increases the orthogonal transform block size to 11.6 ms, for example, as shown in FIG. 2A is used.
[0032]
On the other hand, when the signal is nonstationary, a mode in which the orthogonal transform block size is divided into two or four as compared with the Long Mode is used. More specifically, Short Mode (see FIG. 2B) that divides everything into 4 parts, for example, 2.9 ms, or divides a part into 2 parts, for example, 5.8 ms, and divides the other part into 4 parts. For example, Middle Mode-a (see FIG. 2C) or Middle Mode-b (see FIG. 2D) of 2.9 ms is used. Thus, by setting various time resolutions, it is possible to adapt to actual complex input signals.
[0033]
Obviously, when the constraint on the circuit scale or the like is small, the actual input signal can be more appropriately processed by making the division of the orthogonal transform block size more complicated. The block size as described above is determined by the block determination circuits 109, 110, and 111. Information on the determined block size is supplied to the MDCT circuits 103, 104, and 105 and the bit allocation calculation circuit 118, and the output terminal 113. , 115 and 117.
[0034]
Next, the bit allocation calculation circuit 118 will be described in detail with reference to FIG. Spectral data or MDCT coefficients on the frequency axis from the MDCT circuits 103, 104, and 105, and block size information from the block determination circuits 109, 110, and 111 are supplied to the energy calculation circuit 302 via the input terminal 301. The energy calculation circuit 302 calculates the energy for each unit block, for example, by calculating the sum of the amplitude values in the unit block. Instead of the energy calculation circuit 302, a configuration for calculating peak values, average values, and the like of amplitude values may be provided, and bit allocation processing may be performed based on calculated values such as peak values, average values, and the like of amplitude values. .
[0035]
An example of the output of the energy calculation circuit 302 is shown in FIG. In FIG. 4, the spectrum SB of the total value for each band is indicated by a vertical line segment with a circle at the tip. Here, the horizontal axis represents frequency and the vertical axis represents signal intensity. In FIG. 4, the number of divisions per unit block is 12 blocks (B1 to B12), and only the spectrum of B12 is denoted by the symbol “SB” in order to avoid complicated illustration.
[0036]
The energy calculation circuit 302 performs a process of determining a scale factor value that is normalization information indicating the block floating state of the unit block. Specifically, for example, some positive values are prepared in advance as candidates for the scale factor value, and among them, the value of the spectrum data in the unit block or the absolute value of the MDCT coefficient is greater than the maximum value. The smallest one is adopted as the scale factor value of the unit block. The scale factor value candidates may be numbered using, for example, several bits in a form corresponding to the actual values, and the numbers may be stored in a ROM (Read Only Memory) or the like (not shown). At this time, the scale factor value candidates are defined so as to have values at intervals of 2 dB, for example, in numerical order. A number assigned to a scale factor value adopted for a certain unit block is used as sub information, and is used as scale factor information for the unit block.
[0037]
The output of the energy calculation circuit 302, that is, each value of the spectrum SB is sent to the convolution filter circuit 303. The convolution filter circuit 303, for example, includes a plurality of delay elements that sequentially delay input data, a plurality of multipliers that multiply the output from these delay elements by a filter coefficient (weighting function), and a sum of outputs from the multipliers. And a sum adder. The convolution filter circuit 303 performs a convolution process such that the spectrum SB is multiplied by a predetermined weighting function and added in order to consider the influence on the masking of the spectrum SB. By this convolution processing, the sum total of the portions indicated by dotted lines in FIG. 4 is calculated.
[0038]
Returning to FIG. 3, the output of the convolution filter circuit 303 is supplied to the arithmetic unit 304. The computing unit 304 is further supplied from the (n-ai) function generation circuit 305 with an allowable function (a function expressing the masking level). The arithmetic unit 304 calculates a level α corresponding to an allowable noise level in the region convolved by the convolution filter circuit 303 according to the tolerance function. Here, the level α corresponding to the allowable noise level (allowable noise level) is set to an allowable noise level for each band of the critical band by performing a reverse convolution process, as will be described later. The level. The calculated value of the level α is controlled by increasing or decreasing the allowable function.
[0039]
That is, the level α corresponding to the allowable noise level can be obtained by the following equation (1), where i is a number given sequentially from the lowest band of the critical band.
[0040]
α = S− (n−ai) (1)
[0041]
In equation (1), n and a are constants, a> 0, S is the intensity of the spectrum subjected to convolution processing, and (n−ai) in equation (1) is an allowable function. As an example, n = 38 and a = 1 can be set.
[0042]
The level α calculated by the calculator 304 is transmitted to the divider 306. The divider 306 performs a process of deconvolution of the level α, and as a result, generates a masking spectrum from the level α. This masking spectrum becomes an allowable noise spectrum. In general, when performing the inverse convolution process, it is necessary to perform a complicated operation. However, in one embodiment of the present invention, the simplified convolution unit 306 is used to perform the inverse convolution. ing. The masking spectrum is supplied to the synthesis circuit 307. Further, data indicating a minimum audible curve RC as described later is supplied from the minimum audible curve generating circuit 312 to the synthesis circuit 307.
[0043]
The combining circuit 307 generates a masking spectrum by combining the masking spectrum that is the output of the divider 306 and the data of the minimum audible curve RC. The generated masking spectrum is supplied to the subtracter 308. Further, the output of the energy detection circuit 302, that is, the spectrum SB for each band is supplied to the subtracter 308 after the timing is adjusted by the delay circuit 309. The subtracter 308 performs a subtraction process based on the masking spectrum and the spectrum SB.
[0044]
As a result of such processing, the portion of the spectrum SB for each block below the masking spectrum level is masked. FIG. 5 shows an example of masking. It can be seen that the portion of the spectrum SB below the masking spectrum level (denoted as MS) is masked. In order to avoid complication of illustration, in FIG. 5, only “B” is given to the spectrum, and “MS” is given to the level of the masking spectrum only in B12.
[0045]
If the absolute noise level is below the minimum audible curve RC, the noise cannot be heard by humans. Even if the coding is the same, the minimum audible curve differs depending on, for example, the reproduction volume during reproduction. However, in an actual digital system, for example, there is not much difference in how music data enters the 16-bit dynamic range. For example, if quantization noise in the frequency band that is most audible near 4 kHz, for example, cannot be heard, It is considered that quantization noise below the level of the minimum audible curve cannot be heard in other frequency bands.
[0046]
Therefore, for example, when the system is used such that the noise around the 4 kHz word length of the system cannot be heard, an allowable noise level can be obtained by synthesizing the minimum audible curve RC and the masking spectrum MS. The allowable noise level is a portion indicated by hatching in FIG. Here, the level of 4 kHz of the minimum audible curve is set to the lowest level corresponding to 20 bits, for example. In FIG. 6, SB is shown as a horizontal solid line in each block, and MS is shown as a horizontal dotted line in each block. However, in order to avoid complication of illustration, in FIG. 6, only the spectrum of B12 is denoted by “SB” and “MS”. In FIG. 6, the signal spectrum SS is indicated by a one-dot chain line.
[0047]
Returning to FIG. 3, the output of the subtracter 308 is supplied to the allowable noise correction circuit 310. The allowable noise correction circuit 310 corrects the allowable noise level at the output of the subtractor 308 based on, for example, data of an equal loudness curve. That is, the allowable noise correction circuit 310 calculates the allocation bit for each unit block based on the various parameters such as masking and auditory characteristics described above. The output of the allowable noise correction circuit 310 is output as final output data of the bit allocation calculation circuit 118 via the output terminal 311. Here, the equal loudness curve is a characteristic curve relating to human auditory characteristics, for example, the sound pressure of sound at each frequency that is heard at the same magnitude as a pure tone of 1 kHz is obtained and connected by a curve. Also called sensitivity curve.
[0048]
Further, this equal loudness curve draws the same curve as the minimum audible curve RC shown in FIG. In this equal loudness curve, for example, even if the sound pressure is 8 to 10 dB lower than 1 kHz near 4 kHz, it can be heard as large as 1 kHz. It doesn't sound the same size. Therefore, if noise exceeding the level of the minimum audible curve RC (allowable noise level) has frequency characteristics along the equal loudness curve, the noise can be prevented from being heard by humans.
[0049]
It can be seen that correcting the allowable noise level in consideration of the equal loudness curve is suitable for human auditory characteristics. As described above, the bit allocation calculation circuit 118 obtains data obtained by processing the orthogonal transform output spectrum as the main information with the sub information, and the scale factor indicating the block floating state and the word length indicating the tone as the sub information. Based on these pieces of information, the adaptive bit encoding circuits 106, 107, and 108 in FIG. 1 perform requantization to generate highly efficient encoded data according to the encoding format.
[0050]
Returning to FIG. 1, the normalization information changing circuit 119 will be described. As described above, by manipulating the scale factor information determined by the energy calculation circuit 302, for example, level adjustment every 2 dB can be performed. The normalization information change circuit 119 generates values related to the change of the scale factor information, and supplies the generated values to the computing units 120, 121, and 122, respectively. The arithmetic unit 120 adds the values supplied from the normalization information changing circuit 119 to the scale factor information in the encoded data supplied from the adaptive bit allocation encoding circuits 106, 107, and 108, respectively. To do. However, when the value output from the normalization information change circuit 119 is negative, the arithmetic units 120, 121, 122 act as subtractors. The addition result at this time is limited so as to be within the range of the numerical value of the scale factor defined in the format.
[0051]
Note that when the normalization information change circuit 119 outputs the same value for all unit blocks as a value to be added to the scale factor information, level adjustment processing is performed, but the normalization information change circuit 119 If different values are output for each block, for example, filter processing or the like can be realized. When performing filter processing or the like, the normalization information changing circuit 119 outputs a set of a value to be added to the scale factor information and a unit block number related to the scale factor information to which the value is to be added. . The normalization information adjustment process as described above can also be realized in the case of decoding described later.
[0052]
Next, the encoding format of the highly efficient encoded data will be described with reference to FIG. Numerical values 0, 1, 2,..., 211 shown on the left side represent the number of bytes, and in this example, 212 bytes are used as a unit of one frame. The block size information of each band determined by the block determination circuits 109, 110, and 111 in FIG. Information on the number of unit blocks to be recorded is recorded at the position of the next first byte. For example, as the frequency becomes higher, the bit allocation is set to 0 by the bit allocation calculation circuit 118 and recording is often unnecessary. Therefore, by setting the number of unit blocks so as to cope with such a situation, Many bits are distributed to the middle and low range, which has a great influence on hearing. At the same time, the number of unit blocks in which bit assignment information is double-written and the number of unit blocks in which scale factor information is double-written are recorded at the position of the first byte.
[0053]
Double writing is a method of recording the same data as data recorded at a certain byte position in another location for error correction. Increasing the amount of data that is written twice increases the strength against errors, but the amount of data that can be used for spectrum data increases as the amount of data that is written twice is reduced. In an example of this encoding format, by setting the number of unit blocks for performing double writing independently for each of bit allocation information and scale factor information, it is used for recording the strength against errors and spectrum data. The number of bits is made appropriate. For each piece of information, the correspondence between the number of codes and unit blocks within the prescribed bits is determined in advance as a format.
[0054]
An example of recorded contents in 8 bits at the position of the first byte is shown in FIG. Here, the first 3 bits are used as information on the number of unit blocks to be actually recorded, the subsequent 2 bits are used as information on the number of unit blocks in which bit assignment information is written twice, and the last 3 bits are used as information. This is information on the number of unit blocks in which scale factor information is double-written.
[0055]
In FIG. 8, the bit allocation information of the unit block is recorded at the position from the second byte. For recording bit allocation information, for example, 4 bits are used per unit block. Thereby, bit allocation information corresponding to the number of unit blocks recorded in order from the 0th unit block is recorded. After the bit allocation information data, the scale factor information of each unit block is recorded. For recording scale factor information, for example, 6 bits are used per unit block. Thereby, the scale factor information for the number of unit blocks recorded in order from the 0th unit block is recorded.
[0056]
After the scale factor information, the spectrum data in the unit block is recorded. The spectrum data is recorded in order from the 0th unit block for the number of unit blocks to be actually recorded. Since how many pieces of spectrum data exist for each unit block is determined in advance by the format, it is possible to take correspondence of data by the above-described bit allocation information. Note that recording is not performed for a unit block whose bit allocation is 0.
[0057]
After the spectrum information, the above-described double writing of the scale factor information and the double writing of the bit allocation information are performed. This double-write recording method is the same as the above-described recording of scale factor information and bit allocation information except that the correspondence of the number corresponds to the double-write information shown in FIG. . In the last byte, that is, the 211th byte, and the position that is one byte before that, that is, the 210th byte, the information of the 0th byte and the 1st byte is written twice. These two-byte double writing is defined as a format, and the double writing recording amount cannot be set variable like the double writing of the scale factor information and the double writing of the bit allocation information.
[0058]
Next, a decoding process for decoding high-efficiency encoded data will be described. An example of the configuration of the decoding processing system is shown in FIG. The highly efficient encoded data is supplied to the computing unit 710 via the input terminal 707. Also, block size information used in the encoding process, that is, data equivalent to the output signals of the output terminals 113, 115, and 117 in FIG. 1 is supplied to the input terminal 708. Also, the normalization information change circuit 709 generates a value to be added to or subtracted from the scale factor information of each unit block.
[0059]
The arithmetic unit 710 is further supplied with numerical data from the normalization information change circuit 709. The computing unit 710 adds the numerical data supplied from the normalization information changing circuit 709 to the scale factor information in the supplied high efficiency encoded data. However, when the numerical data supplied from the normalization information change circuit 709 is a negative number, the calculator 710 acts as a subtracter. The output of the arithmetic unit 710 is supplied to an adaptive bit allocation decoding circuit 706 and an output terminal 711.
[0060]
The adaptive bit allocation decoding circuit 706 performs processing for releasing the bit allocation with reference to the adaptive bit allocation information for each of the high frequency band, the mid frequency band, and the low frequency band. The output of the adaptive bit allocation decoding circuit 706 for each of the high band, middle band, and low band is supplied to inverse orthogonal transform circuits 703, 704, and 705. The inverse orthogonal transform circuits 703, 704, and 705 perform inverse orthogonal transform processing on the supplied data. Thereby, the signal on the frequency axis is converted into a signal on the time axis. The partial band signals on the time axis, which are the outputs of the inverse orthogonal transform circuits 703, 704, and 705, are synthesized by the band synthesis filters 701 and 702, and decoded into full band signals. As the band synthesis filters 701 and 702, for example, IQMF (Inverse Quadrature Mirror filter) can be used.
[0061]
By manipulating the scale factor information by addition or subtraction by the arithmetic unit 710, it is possible to adjust the level of reproduction data, for example, every 2 dB. For example, the same numerical value is output from the normalization information changing circuit 709, and the level adjustment in units of 2 dB is performed for all unit blocks by the process of uniformly adding or subtracting the numerical value to the scale factor information of all unit blocks. It is possible to do.
[0062]
Further, for example, independent numerical values are output for each unit block from the normalization information change circuit 709, and the level adjustment for each unit block is performed by adding or subtracting these numerical values to the scale factor information of each unit block. As a result, a filter function can be realized. More specifically, the normalization information changing circuit 709 outputs a unit block and the unit block by a method such as outputting a set of a unit block number and a value to be added to or subtracted from the scale factor information of the unit block. The scale factor information is associated with the value to be added or subtracted. Note that the scale factor information generated as a result of addition or subtraction by the arithmetic unit 710 is limited so that the corresponding scale factor value falls within the range defined by the format of the highly efficient encoded data.
[0063]
The scale factor value that has been subjected to the level adjustment of the unit block by the arithmetic unit 710 is used for only the level adjustment of the decoded signal by being used in the decoding process of the adaptive bit allocation decoding circuit 706. For example, the scale factor value is read from the recording medium on which the encoded information is recorded, the adjusted scale factor value is output to the output terminal 711, and the scale factor value recorded on the recording medium is output. It is also possible to change to an adjusted value. The information on the recording medium can be changed as necessary. As a result, the level information of the recording medium can be changed with a very simple system.
[0064]
In the above description, the scale factor information changing process is performed in both the encoding circuit and the decoding circuit. On the other hand, even when the scale factor information changing process is performed only in the decoding circuit, functions such as level adjustment and filtering can be sufficiently obtained as a result of the changing process.
[0065]
Next, a time unit for performing the processing in the above-described high efficiency encoding will be described. 1 is supplied with audio PCM samples. In the MDCT processing performed by the MDCT circuits 103, 104, and 105 performed after the input, the number of samples for performing so-called orthogonal transform processing is defined. It becomes a unit and is repeatedly processed.
[0066]
Here, 1024 PCM samples input from the input terminal 100 are output from the MDCT circuits 103, 104, and 105 as 512 MDCT coefficients or spectrum data. Specifically, 1024 PCM samples input from the input terminal 100 are converted into 512 high-frequency samples, 512 low-frequency samples, and 256 mid-frequency samples by the band division filter 101. Thereafter, 256 low-frequency samples from the band division filter 102 become 128 low-frequency spectrum data by the MDCT circuit 105, and 256 middle-frequency samples from the band division filter 102 are obtained by the MDCT circuit 104. The 128 high-frequency spectrum data are converted into 128 high-frequency spectrum data, and the 512 high-frequency samples from the band division filter 101 are converted into 256 high-frequency spectrum data by the MDCT circuit 103. In this way, a total of 512 pieces of spectrum data are created from 1024 PCM samples. These 1024 PCM samples are a time unit for performing the above-described high-efficiency encoding once, and are the 212-byte high-efficiency encoded data shown in FIG. 7, that is, one frame.
[0067]
As described above, one frame is composed of, for example, 1024 PCM samples, but in the MDCT processing by the MDCT circuits 103, 104, and 105 in FIG. Arise. The relationship between the PCM sample and the frame will be described with reference to FIG. As shown in FIG. 10, for example, when 1024 PCM samples from nth to n + 1023 are processed in the Nth frame, 1024 PCMs from n + 512th to n + 1535th in the N + 1th frame. Samples are processed, and in the (N + 2) th frame, 1024 PCM samples from the (n + 1024) th to the (n + 2047) th are processed. As described above, one frame has an overlap between adjacent sound frames and 512 PCM samples. That is, when processing is performed in this manner, one frame of high-efficiency encoded information is obtained by processing 1024 PCM samples, but considering overlap with adjacent frames, 512 PCM samples. That would be considerable.
[0068]
FIG. 10 shows the correspondence with the frame in the middle of the PCM sample. With respect to the start point of the PCM sample, for example, 512 PCM samples of 0 data are assumed at a stage before the start point. It is assumed that the PCM sample of 0 data is overlapped with the virtual frame before the first frame and processed. Also, in the last frame, 512 zero-data PCM samples are assumed after the end of the sample sequence, and these 512 zero-data PCM samples overlap the virtual frames after the last frame. Shall be processed.
[0069]
Next, a method for processing the above-described encoding or decoding method as software on a so-called personal computer will be described. As processing on a personal computer, a high-efficiency encoded data file is created on the hard disk by performing high-efficiency encoding mainly on the PCM data file on the hard disk, or a high-efficiency encoded data file on the hard disk is created. It is conceivable to create a PCM data file on the hard disk by performing decryption processing. At this time, one piece of music usually corresponds to one file.
[0070]
As a specific example, screen display, operation method, processing process, etc. in software using a GUI (Graphical User Interface) in a so-called personal computer will be described with reference to FIG. FIG. 11 shows an example of screen display on the personal computer of the encoding and decoding software. The software first selects a directory for PCM data and high efficiency encoded data. Reference numeral 801 denotes a display section for the directory path of the PCM data file. In this example, a directory named PCMDATA in the C drive is currently selected. Reference numeral 803 denotes a display operation unit that displays the file structure in the directory indicated by the display unit 801 and can perform directory movement, drive movement, file selection, and the like. In this example, it can be seen that a directory named tmp exists under the directory indicated by the current display unit 801.
[0071]
In addition, it is assumed that the display “..” indicates a directory one level above. Further, the six files below tmp indicate PCM data files. Also, [-c-] and [-d-] below it indicate a movable drive. Whether the displayed item is a directory, a drive, or PCM data can be determined by a displayed character string or a so-called icon added to the side of the character string.
[0072]
The display section of the directory and drive can move the current directory position to the double-clicked position by associating the mouse pointer with the character string position and double-clicking. In this example, for example, when double-clicking is performed at the location of tmp, the display on the display unit 801 is C: ¥ PCMDATA ¥ tmp, and the display operation unit 803 indicates the state of the file under tmp and the movable drive. It comes to be. In this way, by repeatedly double-clicking the drive name or directory name, it is possible to move to the desired directory position for the PCM data file.
[0073]
Reference numeral 802 denotes a display unit that displays a directory position for high-efficiency encoded data. In the illustrated example, a directory named ENCODEDATA of the C drive is selected. Reference numeral 804 denotes a display operation unit that displays a file structure in the directory indicated by the display unit 802 and can perform directory movement, drive movement, file selection, and the like. In this example, it is indicated that neither a file nor a directory exists under the current directory shown on the display unit 802. The operations in the display operation unit 804 and the correspondence with the display unit 802 are the same as those in the display operation unit 803 and the display unit 801. The display operation unit 804 selects a directory for high-efficiency encoded data. Can do.
[0074]
Reference numeral 805 denotes a button for executing high-efficiency encoding. By clicking here, the PCM data file selected by the display operation unit 803 is sequentially encoded with high-efficiency, and the directory of the directory indicated by the display unit 802 is displayed. A high-efficiency encoded file is created below. The actual processing flow will be described with reference to FIG.
[0075]
In the state shown in FIG. 12A, the display operation unit 803 in FIG. pcm, dataA. pcm, dataB. Three PCM files of pcm are selected and highlighted. When the button 805 in FIG. 11 is clicked here, these three files are sequentially encoded with high efficiency. In the case of normal high-efficiency encoding processing, the order of files to be processed is not particularly problematic.
[0076]
The state shown in FIG. 12B shows a display screen during execution of the high-efficiency encoding process, and the progress of the encoding process can be recognized in the form of a bar graph. Although not shown here, a means for canceling the process in the form of a button may be provided. FIG. 12C shows a state where the high-efficiency encoding process has been completed for all the selected files. The operation display unit 804 in FIG. 11 includes three high-efficiency encoded data files, data2enc. data, dataAenc. data, dataBenc. dat is displayed. The file name of the high-efficiency encoded data file after processing is arbitrary, but here is the so-called extension part of the PCM file name to be processed. The name of the part from which pcm is removed is enc. A file name automatically added with dat is adopted.
[0077]
Next, the button 807 will be described. This button 807 is used to continuously handle high-efficiency encoding processing of a plurality of files as a data string. As described with reference to FIG. 12B, when files are continuously processed, the process according to FIG. 1 and the data relationship shown in FIG. 10 are performed for each file. For this reason, as described above, all files to be processed are processed in consideration of the assumption of 512 zero-data PCM samples at the start point and the assumption of zero-data PCM samples at the end point. . Normally, this method does not pose a problem if the music is independent for each file, but if it is separate as a music but continuous as PCM data, by performing a highly efficient encoding process, Continuity will be lost.
[0078]
An example of this is shown in FIG. pcm, dataB. The case where pcm is continuous PCM data will be described with reference to FIGS. 13A, 13B, and 13C. In FIG. 13A, dataA. PCM data at the end point of pcm and dataB. It shows that the PCM data at the starting point of pcm is continuous.
[0079]
Further, it is assumed that the final part of the frame division for performing the high-efficiency encoding processing described with reference to FIG. 10 or the like is in a state such as N and N + 1 in FIG. 13A. At this time, dataA. FIG. 13B shows the processing of the final part of pcm. That is, the (N + 1) th frame is the final frame, but the data after the dividing point in FIG. 13A is data of another file, so the data after the dividing point is not used and 0 data is obtained for the fractional portion. To process.
[0080]
In contrast, dataB. For the data of the start point of pcm, the process shown in FIG. 13C is performed. That is, since the data before the division point in FIG. 13A is data of another file, the data before the division point is not used, and the 1024 PCM data of the first frame is 512 zero data and 512 dataB. . It consists of data of the starting point of pcm.
[0081]
At this time, the dataA. The frame allocation for processing pcm and the dataB. The frame allocation for processing pcm is different. In addition, since zero data is inserted as a fraction, continuity is lost. That is, dataA. pcm and dataB. When pcm is reproduced continuously, it becomes a continuous sound. data and dataBenc. When dat is decoded and continuously reproduced, the sound is cut off.
[0082]
On the other hand, an example in which the data string is processed in a continuous form by clicking the button 807 in FIG. 11 will be described with reference to FIGS. 14A, 14B, and 14C. As shown in FIG. 14A, the file division points and dataA. The pcm processing frame allocation and the like are in the same state as in FIG. 13A. FIG. 14B shows dataA. This shows the state of the final frame of pcm, but unlike FIG. 13B, data B.R is not filled with 0 data in the data outside the dividing point. The data of pcm is adopted.
[0083]
FIG. 14C shows dataB. Although the top frame allocation of pcm is shown, as shown in FIG. 13C, instead of filling the frame with the start point of the file and filling 0 data, dataA. As a frame division process that maintains continuity with the frame division of pcm, dataB. For data outside the starting point of pcm, dataA. The data of pcm is adopted. That is, N + 2 when considering the frame division in FIG. This is the first frame of pcm. By processing in this way, continuity is maintained between the two files even in high-efficiency encoded data, and dataAenc. data and dataBenc. When dat is decoded and continuously reproduced, no sound interruption occurs.
[0084]
As shown in FIG. 13A, FIG. 13B, and FIG. 13C described above, the processing in the case of performing the encoding processing is shown in the flowchart of FIG. 15, and the encoding processing is performed as shown in FIGS. 14A, 14B, and 14C. The process in the case of performing is shown in the flowchart of FIG.
[0085]
In the first step S1 of FIG. 15, a reading buffer for 1024 points is prepared. Next, the number i of the file to be processed is set to 0 (step S2). In step S3, it is determined whether there is an i-th file to be processed. If there is no file, the process ends (step S4).
[0086]
If the i-th file exists, in step S5, a process of filling zero data as 512 points of data in the first half of the read buffer is performed. Next, the i-th read file (PCM file) is opened (step S6), and the i-th write file (encoded file) is opened (step S7). Data is read from the read encoding into the latter half 512 points of the buffer (step S8).
[0087]
In step S9, the read data amount is acquired, and the read position is updated. In step S10, it is determined whether the amount of read data is less than 512 points. If the read data amount is less than 512 points, in step S11, zero data is packed as data of the difference between the read buffer 512 points and the read data amount.
[0088]
When the read data amount is 512 points in step S10, or following step S11 (zero data filling), encoding processing for one frame is performed in step S12. In step S13, the encoded data is written into the write file.
[0089]
If the result of determination in step S10 is affirmative (if the read data amount is less than 512 points), the i-th read file is closed in step S14, the i-th write file is closed in step S15, and step In S16, i is incremented. Then, the process returns to step S3 (determination of presence / absence of i-th file).
[0090]
If the result of the determination in step S10 is negative (if the read data amount is 512 points), the process of step S17 is performed following step S13. In step S17, the data for the latter half 512 points of the read buffer is shifted to the first half 512 points. Then, the process returns to step S9 (acquisition of read data amount and update of read position).
[0091]
In this way, as shown in FIG. 13, the process applied when the music is independent for each file is performed. Further, a process (FIG. 14) applied when the PCM data is continuous although it is different from the music will be described with reference to the flowchart of FIG.
[0092]
In the first step S21, a reading buffer for 1024 points is prepared. In step S22, i is initialized to 0. In step S23, it is determined whether or not it is the first file (i == 0). In the case of the first file, in step S24, zero data is packed as data for the first half 512 points of the read buffer. Then, the i-th read (PCM) file is opened (step S25), and the i-th write (encoded) file is opened (step S26). In step S27, data is read from the read file to 512 points in the latter half of the buffer.
[0093]
In step S28, the read data amount is acquired, and the read position is updated. In step S29, it is determined whether the amount of read data is less than 512 points. If the read data amount is less than 512 points, it is determined in step S30 whether there is an i + 1 th file to be processed.
[0094]
If it is determined in step S30 that there is no i + 1-th file to be processed, in step S31, zero data is packed as 512 points of data in the read buffer and data of the difference between the read data amounts.
[0095]
If it is determined in step S30 that there is an i + 1 th file to be processed, the i + 1 th read file is opened in step S32. In step S33, 512 points of the reading buffer and the data of the difference amount of the reading data amount are read from the i + 1th file, and the reading position is updated.
[0096]
If the read data amount is 512 points in step S29, following step S31 (zero data filling) or step S33 (reading data from the (i + 1) th file and updating the read position), in step S34, 1 Encoding processing for the frame is performed. In step S35, the encoded data is written into the write file.
[0097]
If the determination result in step S29 is negative (if the read data amount is 512 points), the process of step S36 is performed subsequent to step S35. In step S36, the data for the second half 512 points of the read buffer is shifted to the first half 512 points. Then, the process returns to step S27 (reading data from the read file to the second half 512 points of the buffer).
[0098]
If the determination result in step S29 is affirmative (if the read data amount is less than 512 points), the i-th read file is closed in step S37, and the i-th write file is closed in step S38. Then, when the result of the determination in step S30 is negative (ie, there is no i + 1-th file), the process ends (step S40). On the other hand, if the result of the determination in step S30 is affirmative (that is, there is an i + 1-th file), i is incremented in step S39, and the process in step S36 is performed. Then, the process returns to step S23 (determination of whether or not it is the first file).
[0099]
Next, the button 809 in FIG. 11 will be described. A button 809 is used to analyze file data and determine whether or not to perform the above-described encoding processing in consideration of continuity. Regarding the consideration of continuity, the judgment as to whether or not to consider is usually made by actually listening to the last part and the beginning part of the music file. However, when there are a large number of music files, it is time consuming and inefficient to audition all of them.
[0100]
In general, when a music file is completed with one file and there is no continuity with other files, the data near the beginning or the end of the file becomes silent, and it is zero or near zero. It tends to be a value. On the other hand, in the case of a file that has continuity with other files, the data near the beginning or the end of the file may be other than zero and may have a certain size, that is, a certain volume level. Is expensive. When the button 809 is pressed, based on this feature, the value of data near the beginning or the end of the PCM file selected for encoding is read, and the file is analyzed by analyzing it. A determination is made whether there is a high probability of having continuity with the file.
[0101]
Various analysis methods are conceivable. As the simplest method, for example, the first data and the last data of a file are read one point at a time, and it is determined whether or not they are zero. If the leading data is a value other than zero, the file may have continuity with the last part of another file. If the final data is a value other than zero, the file may have continuity with the head of another file. The user is warned that there may be such continuity. The user can make a reliable judgment by checking the continuity by listening to the file according to the warning.
[0102]
Also, for example, when a file whose final data has a value other than zero is selected as the target of the encoding process, the encoding is performed after the file without performing the above-described warning. It is also possible to automatically perform continuous processing with a file selected to be processed. Similarly, for example, when a file whose leading data has a value other than zero is selected as the target of the encoding process, the above-mentioned warning is not given before the file without performing the above warning. A continuous process may be automatically performed with a file selected to be encoded.
[0103]
In the above description, an example has been described in which the head data and the last data are set one point at a time and the threshold value is set to zero. However, even if it is not actually heard as sound, the beginning data or the last data is a value of a certain size even though there is no continuity with other files due to analog noise, dithering, etc. There are many cases where the data has
[0104]
In order to cope with this problem, a user can freely set a threshold value and the like. The threshold value can be set not only by the level but also by the number of reading points. For example, it is possible to use a method in which the top data and the last data are set at 10 points instead of 1 point, and the sum, average value, etc. of the 10 points are used as threshold values. Alternatively, a method of searching for a difference amount between the final value of a file and the start value of another file is also possible. In this way, by making various threshold settings possible, more appropriate determination can be made.
[0105]
A process of an example of processing when the above-described button 809 is clicked and screen display will be described with reference to FIGS. 17 and 18. First, when a button 809 is clicked, search conditions are set in step S51. This is to set a threshold value. For example, through the setting screen as shown in FIG. 18A, the number of points and the level of data to be read for the first data and the last data are input, and the threshold value calculation method is input. In the example of FIG. 18A, both the top data and the final data are set to be determined based on whether the average value of 10 points exceeds 5. For the calculation method, a sum, a maximum value, etc. can be selected in addition to the average value.
[0106]
Next, in step S52, initialization of variables for sequentially processing the encoding process selection file is performed. Here, this variable is set to i and zero is set. In step S53, whether or not there is a file to be searched is determined by comparing the number of encoding process selection files with the variable i. If the variable i exceeds the number of process selection files in step S53, the process ends (step S54).
[0107]
If it is determined in step S53 that there is a file to be processed, an open process for reading the file is performed in step S55. Thereafter, the head data is read in step S56. The number of data read here is the number of points set in step S51. In step S57, the threshold value set in step S51 is compared with the read data.
[0108]
If the read data exceeds the threshold value in step S57, the warning process in step S58 is performed. In this warning process, for example, a warning message as shown in FIG. 18B is displayed. After the user clicks the OK button being displayed, the process proceeds to the next step. Steps S59, S60, and S61 respectively perform processing corresponding to steps S56, S57, and S58 in the first data for the final data. Similarly, the display of the warning message in FIG. 18C corresponds to the warning display regarding the top data in FIG. 18B.
[0109]
After the processing up to step S60 or S61 is completed, the i-th file is closed in step S62. In step S63, the variable i is incremented, and the process returns to step S53. Thereafter, the user can easily determine whether or not continuity should be considered in the encoding process by auditioning only the file with the warning. In this example, the PCM file is searched before the encoding process. Similarly, the decoding file has a threshold value for the normalization information of the encoded file, the quantization state, and the like. By doing so, it is possible to perform the same processing.
[0110]
Next, the actual encoding process is performed on the basis of the search result. A button 807 in FIG. 11 selects whether processing is performed in the form shown in FIG. 13 or processing is performed in the form shown in FIG. The same applies when the number of continuous files is two or more. The selection of files to be continuously processed is determined by the method shown in FIG.
[0111]
FIG. 19 shows an example of a method for actually setting files to be continuous. FIG. 19 is an operation display screen that appears when the button 807 in FIG. 11 is clicked, and files to be continuously processed are selected on the operation display screen. A display unit 901 displays a file for continuous processing. Here, data2. pcm and data3. An example of continuous processing of pcm is shown. Although the file name can be directly input to the display unit 901, a so-called file structure can be graphically searched and a file can be selected using a button 905. At this time, the order in which the files are selected is reflected in the continuous processing, but the order can be changed in the display unit 901.
[0112]
Further, by using 902, it is possible to support continuous processing of a plurality of files. Similar to the display unit 901, the display unit 903 sets a file to be continuously processed in other combinations. In this example, dataA. pcm, dataB. pcm, dataC. A setting to continuously process pcm is shown. Here, data2. pcm and data3. pcm continuous treatment in the first set, dataA. pcm, dataB. pcm, dataC. Although pcm is the second set, the numerical value of this set is not directly related to the processing result. Reference numeral 904 corresponds to the first set 902. Although two sets are displayed here, such a set can be further set by using 906. Finally, clicking the OK button 907 completes the setting.
[0113]
FIG. 11 will be described again. A button 806 is pressed when the high-efficiency encoded data file selected by the display operation unit 804 is decoded. The processing method, correspondence of display contents, and the like are the same as those by the button 805 at the time of high-efficiency encoding. Also at the time of decoding, it is possible to set the continuous decoding process by using the button 807 as in the case of the above-described high-efficiency encoding continuous process. In the case of continuous decoding processing, the final frame of a certain high-efficiency encoded data file and the first frame of another high-efficiency encoded data file may be set to be decoded. . Reference numeral 808 denotes a button for ending the program.
[0114]
In the method described above, when encoding or decoding multiple files, select whether to process each file independently or to perform processing considering continuity between different files, and select the desired It becomes possible to create a processing file in the form. In addition, it is possible to easily determine whether or not to consider continuity, and it is possible to perform adaptive processing more quickly.
[0115]
【The invention's effect】
In the digital signal processing method according to the present invention described above, it is possible to select encoding in consideration of the continuity of the start and end points between different files and encoding that does not take into account when encoding a plurality of desired files. It is. In addition, according to the present invention, it is easier and more accurate by analyzing the data of the start and end points of a desired plurality of files and selecting the encoding considering continuity and the encoding not considering based on the analysis result. In addition, it is possible to determine whether or not to perform processing in consideration of continuity, and the work efficiency can be greatly improved.
[Brief description of the drawings]
FIG. 1 is a block diagram illustrating an example of a configuration relating to generation of highly efficient encoded data.
FIG. 2 is a schematic diagram for explaining an orthogonal transform block size for each band;
FIG. 3 is a block diagram showing in detail a part of the configuration in FIG. 1;
FIG. 4 is a schematic diagram illustrating an example of a spectrum of a band divided in consideration of a critical band, block floating, and the like.
FIG. 5 is a schematic diagram illustrating an example of a masking spectrum.
FIG. 6 is a schematic diagram for explaining synthesis of a minimum audible curve and a masking spectrum.
FIG. 7 is a schematic diagram illustrating an example of an encoded data format according to an embodiment of the present invention.
8 is a schematic diagram showing details of data of the first byte in FIG. 7; FIG.
FIG. 9 is a block diagram illustrating an example of a configuration related to a digital signal decoding process.
FIG. 10 is a schematic diagram for explaining overlap in each frame in encoded data;
FIG. 11 is a schematic diagram illustrating a specific example of an operation display screen of a system that performs high-efficiency encoding processing and decoding processing on a personal computer.
12 is a schematic diagram showing processing for performing high-efficiency encoding on a plurality of files by the system of FIG.
FIG. 13 is a schematic diagram showing frame correspondence when performing high-efficiency encoding without considering the continuity of two files.
FIG. 14 is a schematic diagram showing frame correspondence when performing high-efficiency encoding in consideration of the continuity of two files.
FIG. 15 is a flowchart showing processing steps when high-efficiency encoding is performed without considering the continuity of two files.
FIG. 16 is a flowchart showing processing steps when high-efficiency encoding is performed in consideration of the continuity of two files.
FIG. 17 is a flowchart showing processing steps for analyzing the top data and the final data.
FIG. 18 is a schematic diagram illustrating a search threshold value input screen and a warning message screen in a processing step of analyzing the top data and the final data.
FIG. 19 is a schematic diagram illustrating a specific example of an operation display screen for selecting a combination of files to be processed in consideration of continuity.
[Explanation of symbols]
101, 102 ... Band division filter, 103, 104, 105 ... Orthogonal transform circuit (MDCT), 109, 110, 111 ... Block decision circuit, 118 ... Bit allocation calculation circuit, 106, 107, 108: adaptive bit allocation encoding circuit, 119: normalized information changing circuit, 120, 121, 122 ... adder, 302 ... energy calculator for each band, 303 ... convolution filter, 304 ... adder, 305 ... function generator, 306 ... divider, 307 ... synthesizer, 308 ... subtractor, 309 ... delay circuit, 310 ... allowable noise correction 701, 702 ... Band synthesis filter (IQMF), 703, 704, 705 ... Inverse orthogonal transform circuit (IMDCT), 706 ... Adaptive bit allocation decoding circuit, 70 ... normalization information change circuit, 710 ... adder, 803 ... display operation unit for PCM data file, 804 ... display operation unit for encoded data file, 807 ... height of multiple files Button for selecting the processing at the time of efficiency encoding, 809 ... button for analyzing the continuity of the file

Claims (5)

An encoding device that blocks a plurality of digital audio files at predetermined lengths and compresses a block-processed digital audio file,
First selection means for selecting two or more digital audio files to be compressed from the plurality of digital audio files in the order of the compression processing ;
Of the two or more digital audio files selected by the first selection means, a block near the end of a digital audio file located in front of an adjacent digital audio file in the order in which the compression processing is performed ; and neighboring blocks beginning of the digital audio files to be located behind the digital audio file which has been said adjacent selected by the first selection means, based on a block that extends over to the adjacent two digital audio files First encoding means for performing encoding processing,
Of the two or more digital audio files selected by the first selection means, a block near the end of a digital audio file positioned in front of an adjacent digital audio file in the order in which the compression processing is performed, or the adjacent Second encoding means for performing encoding processing based on a block in the vicinity of the beginning of the digital audio file located behind the digital audio file to be performed and a block straddling the two adjacent digital audio files When,
Analyzing means for analyzing data of start and end points of the two or more digital audio files selected by the first selection means ;
Marks refer to the analysis result of the analyzing means, Ru and a second selecting means for selecting one of the encoding processing in the encoding process and the second encoding means in the first encoding means Encoding device. The analysis means is
Comparing the volume level of the starting data of the selected digital audio file with a preset threshold;
The encoding apparatus according to claim 1, wherein when the data level of the start point is greater than the threshold value, it is determined that there is continuity with the end point part of another digital audio file. The analysis means is
Comparing the volume level of the end point data of the selected digital audio file with a preset threshold;
2. The encoding apparatus according to claim 1, wherein when the end point data level is larger than the threshold value, it is determined that there is continuity with the head portion of another digital audio file. The second selection means is:
The encoding apparatus according to claim 2 or 3, wherein when the analyzing means determines that there is continuity with the other digital audio file, the encoding process in the first encoding means is selected. A coding method for performing block processing on a plurality of digital audio files for each predetermined length and compressing the block-processed digital audio file,
A first selection step of selecting two or more digital audio files to be compressed from the plurality of digital audio files in the order of the compression processing;
Of the two or more digital audio files selected in the first selection step, a block in the vicinity of the end of the digital audio file located in front of the adjacent digital audio file in the order in which the compression processing is performed; Based on a block near the beginning of the digital audio file located behind the adjacent digital audio file selected in the first selection step, and a block straddling the two adjacent digital audio files. A first encoding step for performing an encoding process;
Of the two or more digital audio files selected in the first selection step, a block near the end of a digital audio file located in front of an adjacent digital audio file in the order in which the compression processing is performed, or the adjacent A second encoding step for performing an encoding process based on a block near the beginning of the digital audio file located behind the digital audio file to be performed and a block straddling the two adjacent digital audio files; When,
An analysis step for analyzing data of start and end points of the two or more digital audio files selected in the first selection step;
Encoding comprising a second selection step of selecting one of the encoding process in the first encoding step and the encoding process in the second encoding step with reference to the analysis result of the analysis step Method.

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