JP2000151414A - Digital audio encoding device/method and recording medium recording encoding program - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide an encoding device whose constitution can be simplified and whose encoding processing of a digital audio signal can be executed at high speed by providing a sub-band division part setting all audio data of a sub-band pertaining to the specified and above frequency area of a frequency which is N times larger than that of the frequency after up-sampling. SOLUTION: An up-sampling part 11 generates PCM data sampled by a sampling frequency which is N times larger than a prescribed sampling frequency from input PCM data sampled by the prescribed sampling frequency based on up-sampling information N given from outside. A sub-band division part 12 sets all the samplings of a band pertaining to a frequency area which is 1/(2×N) of/above the up-sampled sampling frequency to zero. All samples pertaining to sub-band data are set to zero and are outputted to a bit allocation part 14.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital audio encoding apparatus which compresses and encodes an audio signal used in the field of multimedia technology such as personal computers and broadcasting to generate a digital audio encoded signal. The present invention relates to a method and a recording medium on which the encoding program is recorded.

[0002]

2. Description of the Related Art In recent years, with the progress of digital technology,
The performance of personal computers (hereinafter abbreviated as "PCs") has been significantly improved. For this reason, in the personal computer, the MPEG standard (ISO / ISO
It has become possible to reproduce a bit stream based on IEC 11172-3, 13818-3), and to carry out encoding based on the MPEG standard. In the high-efficiency compression technology represented by the MPEG standard, a digital audio signal of a sub-band encoding method in which an original signal of an audio signal is divided into bands and each band signal is encoded. Further, according to the MPEG audio standard, the sampling frequency of a digital audio signal is 32 KH.
z, 44.1 KHz and 48 KHz are specified. On the other hand, in the field of multimedia technologies such as personal computers and broadcasting, PCM data sampled at a sampling frequency of 22.05 KHz or 11.025 KHz has been generally used as a digital audio signal. Therefore, in order to execute data processing based on the above-mentioned MPEG audio standard in a personal computer, the PCM data is converted to 44.
It is necessary to convert the data to data up-sampled to 1 KHz before encoding.

Hereinafter, a conventional digital audio encoding device will be described in detail with reference to FIG. In the following description, one channel (monaural) PCM data sampled at 22.05 KHz is converted to 44.1K.
A case where a digital audio signal based on the MPEG1 audio standard is generated by upsampling to Hz will be described. FIG. 3 is a block diagram showing a configuration of a main part of a conventional digital audio encoding device. Referring to FIG. 3, this conventional digital audio encoding apparatus is an MPE using the MPEG1 audio standard.
It is divided into a G audio encoding unit 30 and an input data generation unit (not shown) which is provided at the preceding stage and generates input PCM data from an original signal of the audio signal. further,
The conventional digital audio encoding device is provided with an up-sampling unit 31 for up-sampling input PCM data from an input data generating unit, and a filter processing unit 32 for filtering PCM data from the up-sampling unit 31. ing. In this conventional digital audio encoding apparatus, the input data generation unit sets how many times the sampling frequency of the original PCM data is up-sampled, and sets up-sampling information N (N is a factor of 2). Output to the upsampling unit 31 and the filter processing unit 32. In this description, up-sampling is performed from 22.05 KHz to 44.1 KHz.

[0004] The upsampling section 31 provides upsampling information N (=) provided from an input data generating section.
2), a predetermined sampling frequency (22.05
KM), and generates PCM data sampled at N times the sampling frequency (44.1 KHz) from the input PCM data sampled at KHz. In particular,
The up-sampling unit 31 interpolates the average value of the adjacent input PCM data to obtain the Ps after the up-sampling.
The CM data is output to the filter processing unit 32. The filter processing unit 32 performs a filtering process on the PCM data up-sampled by the up-sampling unit 31 to remove aliasing noise, and outputs the result to the MPEG audio encoding unit 30.

Here, with reference to FIG. 4, the interpolation processing in the up-sampling section 31 in the conventional digital audio encoding device will be described in detail. FIG.
FIG. 4 is an explanatory diagram illustrating an interpolation process in the upsampling unit illustrated in FIG. 3. First, in order to encode an audio signal based on the MPEG1 standard, 1 channel (monaural) audio signal requires 1152 pieces of PCM data, and 2 channel (stereo) audio signal is twice as large as 2304 pieces of data. PCM data is required, and each of them is referred to as one frame of PCM data. In this description, the case of one channel is described, so that the up-sampling unit 31 has 1152 PCMs.
The up-sampled PCM data for each data is passed through a filter processor 32 to an MPEG audio encoder 30.
Output to Therefore, in the up-sampling unit 31,
As shown in FIG. 4, upsampled PCM data interpolated by an average value of two adjacent input PCM data in the original input PCM data is generated. here,
Since the up-sampling unit 31 performs double up-sampling, as shown in FIG.
Total 57 of data and PCM data at the head of next frame
1152 pieces of PCM data are generated from the seven pieces of PCM data. As described above, the PCM data necessary for encoding is generated by the up-sampling unit 31, and the PCM data includes aliasing noise generated when the subsequent MPEG audio encoding unit 30 performs frequency division. Therefore, the above-described filter processing unit 32 (FIG. 3) performs band limitation to remove aliasing noise.
The MPEG audio encoder 30 (FIG. 3) converts the band-limited 1152 pieces of PCM data to 44.1 kHz.
Is encoded as data sampled by.

Returning to FIG. 3, a conventional digital audio encoding device including an MPEG audio encoding unit 30 is known, for example, from Japanese Patent Application Laid-Open No. 9-134200. First, a human psychoacoustic model used in the MPEG audio encoding unit 30 will be described. In other words, in the compression encoding of an audio signal according to the MPEG standard, a perceptual code that reduces (compresses) the amount of code (data) by omitting information with low sensitivity and using details of the human sense of receiving the audio signal. Is adopted. More specifically, audio signals that cannot be heard are removed by using the masking phenomenon and the critical band in the auditory characteristics, and only necessary audio signals are encoded and bits are allocated, so that the bits can be encoded with fewer bits than the original audio signal. Even if the sound quality is changed, the same sound quality as the original sound can be obtained. Here, the masking phenomenon means that one audio signal is masked by another audio signal due to interference between audio signals,
It is a phenomenon that cannot be heard at all. Further, the critical band is a kind of unit for dividing the frequency of a sound by a human, and is generally divided into 24 bands. The higher the frequency of these bands, the greater the width of the band on a log scale (logarithmic unit). Therefore, in the critical band, it is more difficult to distinguish high frequency signals than low frequency signals. Since the MPEG audio encoding unit 30 allocates bits using the above-described auditory characteristics, a signal-to-noise ratio (SNR) and a signal-to-mask level ratio (SMR) are obtained. Noise ratio (MN
R) is calculated. Here, the mask level is the minimum signal level that cannot be heard. Therefore, there is no need to assign bits to signals below this mask level.

[0007] In order to perform the above-described perceptual encoding on the band-limited PCM data from the filter processing unit 32, the MPEG audio encoding unit 30 includes a sub-band division unit 33, a bit allocation information generation unit 34, It comprises a bit allocating unit 35 and a multiplexing unit 36. The sub-band division unit 33 converts the PCM data in the time domain from the filter processing unit 32 into sub-band data in the frequency domain divided into 32 equal sub-bands. The subband data includes 12 (in the case of layer I) or 36 (in the case of layer II) samples in each subband. Further, the sub-band dividing unit 33 obtains a scale factor indicating a sound pressure level at the time of reproduction from the input PCM data and encodes the scale factor. Since the number of scale factors is 64 in total, the number of bits required for encoding this information is 6 bits. Furthermore, the encoding method of this information differs for each layer,
Then, the maximum value of the twelve samples existing in each band (sub-band) is obtained, and it is determined whether the maximum value matches the obtained maximum value.
A slightly larger value is used as the scale factor of the band.
On the other hand, in Layer II, since there are three scale factors in each band, the similarity of each scale factor is examined to determine how many of the three scale factors are to be coded. That is, the number of scale factors to be encoded differs depending on the range of the difference value between adjacent scale factors. Thus, unlike Layer I, Layer II requires additional information to select the scale factor, but the scale factor is encoded with two bits.

[0008] The bit allocation information generation unit 34 utilizes the human auditory characteristics to convert the PCM data input from the filter processing unit 32 into the aforementioned signal-to-mask level ratio (SM).
R). The final output value of the bit allocation information generator 34 is used as bit allocation information (SMR value) as a bit allocation reference for each band.
5 is output. Specifically, this SMR value is calculated by a series of nine steps as follows. In the first stage, the audio signal (PCM data) in the time domain is converted into the frequency domain by the fast Fourier transform (FFT), and in the second stage, the sound pressure level in each band is calculated. In a third step, an absolute masking threshold value is calculated, and in a fourth step, voiced and unvoiced sound components of the audio signal are determined. In the fifth step, a masker which is a sound to be masked among voiced sounds is determined, and in the sixth step, a masking threshold value of each band is calculated. In the seventh step, masking threshold values of all bands are calculated, and in the eighth step, the minimum masking threshold value of each band is calculated. In the ninth stage, the SMR value of each band is calculated.

[0009] The bit allocation unit 35 performs bit allocation processing and quantization processing on the sub-band data of each band from the sub-band division unit 33, and outputs the result to the multiplexing unit 36.
In the bit allocation process, the bit allocation information generation unit 3
Based on the SMR value from No. 4, the following series of four steps are repeated to determine the number of bits allocated to each band. In the first stage, the initial allocation bits are set to 0, and the second
In the step, an MNR value, which is the above-described mask level-to-noise ratio, is obtained for each band. At this time, the MNR value is calculated by subtracting the SMR value from the SNR value that is the signal-to-noise ratio for each band. In the third stage, a band having the smallest MNR value is searched for among the MNR values obtained for each band, and the number of bits allocated to the band is increased by, for example, one.
In the fourth step, when the increased number of allocated bits does not exceed the required number of allocated bits, the second and third steps are repeated for the remaining other bands. Further, the quantization process is performed by repeating the following four steps following the bit allocation process. In the first stage, the samples in each band are divided by the scale factor to give X.
In the second stage, “A × X + B” (A and B are predetermined table values) is calculated. In the third step, the number of allocated bits obtained from the bit allocation processing among the values calculated in the second step is obtained. In the fourth step, the most significant bit (MSB) of the obtained allocated bit number is inverted to obtain the quantized data. Output as The multiplexing unit 36 adds a header part and the like defined in the MPEG standard to the quantized data thus obtained, and forms and outputs the MPEG audio bit stream.

As described above, in the conventional digital audio encoding device, the upsampled PCM
The data is band-limited by the filter processing unit 32 and MPEG
The MPEG audio bit stream was formed using the audio encoding unit 30.

[0011]

In the conventional digital audio encoding apparatus as described above, in order to remove aliasing noise, a filter processing section is connected in front of the MPEG audio encoding section, and after upsampling, P
It was necessary to limit the band of CM data (filter processing). For this reason, the conventional digital audio encoding device has a problem that the configuration of the device cannot be simplified. Further, in this conventional digital audio encoding device, the MPEG audio encoding unit performs bit allocation processing and quantization processing on subband data that does not include a frequency component. As a result, in the conventional digital audio encoding device,
It was difficult to improve the processing efficiency of the audio signal encoding process, and the encoding process required a great deal of time.

Further, the conventional digital audio encoding apparatus has a problem that the up-sampling process in the up-sampling section is complicated and takes time. More specifically, in the conventional digital audio encoding device, the up-sampling unit uses one frame of P
When generating CM data, input PCM for one frame
In addition to the data, the input PCM data at the head of the next frame is used. Specifically, as shown in FIG.
The 1152th last PCM data was generated by the average value of the 576th input PCM data and the 1st input PCM data of the next frame. For this reason, in this upsampling unit, it is necessary to acquire not only the processing target frame but also the leading input PCM data of the next frame by performing a pre-reading process. Further, this up-sampling unit performs not only the pre-reading process described above, but also a process of moving the first input PCM data pre-read after the end of the up-sampling process to the original frame, returning the read position information by the pre-reading process, and the like. Processing had to be performed separately.

SUMMARY OF THE INVENTION The present invention has been made to solve the above-mentioned problems, and can simplify the structure of the apparatus, reduce the encoding processing of digital audio signals, and perform high-speed digital processing. An object of the present invention is to provide an audio encoding device, an encoding method, and a recording medium on which the encoding program is recorded.

[0014]

According to the present invention, there is provided a digital audio encoding apparatus for up-sampling input digital audio data sampled at a predetermined sampling frequency based on up-sampling information N (N is a factor of 2). Then, an up-sampling unit that generates audio data sampled at N times the sampling frequency and a band division process of dividing the audio data from the up-sampling unit into predetermined subbands are performed. A sub-band division unit is provided for setting all audio data of sub-bands belonging to a frequency region equal to or more than 1 / (2 × N) of the double sampling frequency to a value of zero. With this configuration, the configuration of the device can be simplified, and the encoding process of the digital audio signal can be performed at high speed.

In a digital audio encoding apparatus according to another aspect of the present invention, when the number of data for one frame of the input digital audio data is M (M is a natural number), the up-sampling unit sets M data from the first to M. The first to (M × N−N) th audio data are generated by using the first input digital audio data, and the Mth input digital audio data is converted from the (M × N−N + 1) th to (M × N−N + 1) th. (M × N) th audio data. With this configuration, it is possible to reduce the processing in the upsampling unit, and to perform the encoding processing of the digital audio signal at high speed.

In the encoding method of the digital audio encoding apparatus according to the present invention, input digital audio data sampled at a predetermined sampling frequency is up-sampled based on up-sampling information N (N is a factor of 2). , An upsampling step of generating audio data sampled at N times the sampling frequency, and a band division process of dividing the audio data generated in the upsampling step into predetermined subbands. Sub-band division step of setting all audio data of sub-bands belonging to a frequency region equal to or more than 1 / (2 × N) of the sampling frequency of the sub-band to zero. With this configuration, the configuration of the device can be simplified, and the encoding process of the digital audio signal can be performed at high speed.

According to another aspect of the present invention, there is provided an encoding method for a digital audio encoding apparatus, wherein when the number of data of one frame of the input digital audio data is M (M is a natural number), The first to Mth input digital audio data are used to generate first to (M × N−N) th audio data, and the Mth input digital audio data is generated as (M × N−N + 1). ) -Th to (M × N) -th audio data. With this configuration, it is possible to reduce the processing in the upsampling unit, and to perform the encoding processing of the digital audio signal at high speed.

The recording medium on which the encoding program of the present invention has been recorded up-samples input digital audio data sampled at a predetermined sampling frequency based on up-sampling information N (N is a factor of 2). Performing an up-sampling step of generating audio data sampled at an N-times sampling frequency, and performing a band division process of dividing the audio data generated in the up-sampling step into predetermined subbands; The method includes a subband division step of setting all audio data of subbands belonging to a frequency region equal to or more than 1 / (2 × N) of the sampling frequency to a value of zero. With this configuration, the configuration of the device can be simplified, and the encoding process of the digital audio signal can be performed at high speed.

According to another aspect of the present invention, there is provided a recording medium on which an encoding program according to the invention is recorded, wherein the number of data of one frame of the input digital audio data is M (M is a natural number),
The first to Mth input digital audio data are used to generate first to (M × N−N) th audio data, and the Mth input digital audio data is generated as (M × N−N + 1). ) To (M
× N) th audio data. With this configuration, it is possible to reduce the processing in the upsampling unit, and to perform the encoding processing of the digital audio signal at high speed.

[0020]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS A preferred embodiment showing a digital audio encoding apparatus and an encoding method according to the present invention will be described below with reference to the drawings. In the following description, in order to facilitate comparison with the conventional example, 1-channel (monaural) PCM data sampled at 22.05 KHz is up-sampled to 44.1 KHz, and the MPEG1 audio standard ( ISO /
A digital audio encoding device that performs encoding based on IEC 11172-3) will be described as an example.

<< Embodiment >> FIG. 1 is a block diagram showing a configuration of a main part of a digital audio encoding apparatus according to an embodiment of the present invention. In FIG. 1, the digital audio encoding apparatus according to the present embodiment includes an MPEG audio encoding unit 10 using the MPEG1 audio standard, and an up-sampling unit 11 provided at a preceding stage for up-sampling input PCM data. . Input PCM
The data is data generated by an input data generator (not shown) by pulse code modulating the original signal of the audio signal. The up-sampling information N (N
Is the factorial of 2). The up-sampling information N is set and notified by, for example, an input data generation unit.
This information specifies how many times the sampling frequency of the original PCM data is to be upsampled. By notifying the up-sampling information N to the MPEG audio encoding unit 10, the filter processing unit in the conventional example shown in FIG. 3 can be omitted, and the configuration of the device can be simplified. Can be performed at high speed (details will be described later). In the following description, the upsampling unit 11 sets the sampling frequency to 22.05K.
Since the frequency is up-sampled from Hz to 44.1 KHz, the up-sampling information N is 2.

The up-sampling section 11 converts input PCM data sampled at a predetermined sampling frequency (22.05 KHz) based on up-sampling information N (= 2) supplied from the outside into an N-fold sampling frequency (44). Generate PCM data sampled at .1 KHz). As will be described in detail later, the up-sampling unit 11 generates PCM data after up-sampling by interpolating with an average value of two adjacent input PCM data in frame units, and outputs the PCM data to the MPEG audio encoding unit 10. . Thus, in the digital audio encoding device according to the present embodiment, the configuration of the upsampling unit 11 can be simplified, and the processing load thereof can be reduced.

The MPEG audio encoding unit 10 includes a subband dividing unit 1 connected to the upsampling unit 11.
2 and a bit allocation information generation unit 13, a bit allocation unit 14 connected to the subband division unit 12 and the bit allocation information generation unit 13, and a multiplexing unit 15 connected to the bit allocation unit 14. I have. The sub-band division unit 12 performs sub-band division on the PCM data input from the up-sampling unit 11 to generate sub-band data. Specifically, the sub-band division unit 12 performs band division to divide time-domain PCM data into 32 equal sub-bands using a band division filter, and generates frequency-domain sub-band data sb0 to sb31. .
These subband data sb0 to sb31 are input PC
The sampling frequency (22.5 KHz) of M data is divided into bands in order from the low frequency band, and each subband data sb0 to sb31 has 12 (in the case of layer I) or 36 (in the case of layer II) Includes samples.

Further, the subband division unit 12 uses the up-sampling information N (= 2) to
All samples in the band not included in the CM data are set to zero. In other words, the sub-band division unit 12
Is the sampling frequency (4
All samples in a band belonging to a frequency region of 1 / (2 × N) or more of (4.1 KHz), that is, a frequency region of 11.25 KHz or more, are set to zero. As a result, all the samples belonging to the sub-band data sb16 to sb31 are set to zero and output to the bit allocation unit 14. Further, it is possible to prevent the generation of aliasing noise that occurs when data processing is performed without cutting a frequency component higher than 1/2 of the sampling frequency (22.5 KHz) of the input PCM data. The processing unit (FIG. 3) can be omitted.

The bit allocation information generating section 13 generates bit allocation information (SMR value) based on the PCM data from the upsampling section 11. For details,
The bit allocation information generation unit 13 generates bit allocation information from the PCM data using the human auditory characteristics and outputs the generated bit allocation information to the bit allocation unit 14. The bit allocation unit 14 performs a bit allocation process of allocating bits to the subband data sb0 to sb31 from the subband division unit 12 based on the bit allocation information from the bit allocation information generation unit 13, and further performs a subband (band). The quantized data is quantized by performing a quantization process every time and output to the multiplexing unit 15. Note that the sub-band data sb1
Since 6 to sb31 are set to zero, the bit allocation unit 14 can omit the bit allocation processing and the quantization processing, and can reduce the processing load. The multiplexing unit 15 forms and outputs the quantized data from the bit allocation unit 14 as an MPEG audio bit stream.

The above-mentioned bit allocation information generating section 1
3, the bit allocation unit 14 and the multiplexing unit 15
Are configured based on the MPEG1 audio standard so as to have the same functions as those of the bit allocation information generation unit, bit allocation unit, and multiplexing unit of the conventional example shown in FIG. Also, in the above description, the subband division unit 1
2 describes the configuration in which all the sub-band data sb0 to sb31 are output to the bit allocating unit 14. However, except for the sub-band data sb16 to sb31 in which zero is set, only the sub-band data sb0 to sb15 are output to the bit allocating unit 14. May be output.

Hereinafter, the operation of the digital audio encoding device according to the present embodiment will be described with reference to FIGS. FIG. 2 is an explanatory diagram showing an interpolation process in the upsampling unit shown in FIG. First, in order to encode an audio signal based on the MPEG1 standard, 1152 PCs are used for a one-channel (monaural) audio signal.
M data is required. For a two-channel (stereo) audio signal, 2304 pieces of PCM data are required, which is twice as much, and each of them is called one frame of PCM data. In this description, the case of one channel is described, so that the up-sampling unit 11
The PCM data after upsampling is output to the MPEG audio encoding unit 10 for each piece of PCM data. Further, as described above, the up-sampling unit 11
Interpolation is performed using an average value of two input PCM data adjacent to each other in a frame unit to generate up-sampled PCM data.

More specifically, one of the original input PCM data
When the number of frames is M (M is a natural number), the up-sampling unit 11 uses the up-sampling information N and outputs the first to (M × N−N) based on the first to M-th original input PCM data. ) -Th PCM data is generated. Further, the up-sampling unit 11
Converts the M-th last input PCM data to (M × N−N +
It is repeatedly used as (1) -th to (M × N) -th PCM data. Specifically, as shown in FIG. 2, the up-sampling unit 11 calculates the average value of the first to 576th original input PCM data and the average value of two adjacent input PCM data, and Value of the original adjacent two
Interpolation between two input PCM data, 1st to 11th
It is generated as PCM data after up-sampling up to the 50th. For example, the upsampling unit 11
Uses the first and second original input PCM data as the first and third PCM data, and sets the average value thereof as the second PCM data. Further, as shown in FIG.
The 576th last input PCM data is used for the 1152th PCM data.

As described above, the up-sampling unit 11 of this embodiment uses only one frame of input PCM data to generate PCM data for that frame. Therefore, unlike the conventional example, the upsampling unit 11 of the present embodiment does not need the input PCM data at the head of the next frame, and can perform the upsampling process at high speed. Further, the upsampling unit 11 uses only the last input PCM data to generate the (M × N−N + 1) th to (M × N) th PCM data.
Since the M data is generated, a decrease in sound quality can be suppressed.

Subsequently, the sub-band division unit 12 performs band division for dividing the PCM data from the up-sampling unit 11 into 32 equal sub-bands, generates sub-band data sb0 to sb31, and Output to At this time, the subband dividing unit 12 uses the up-sampling information N (= 2) to
All samples in the band not included in the CM data are set to zero. Next, the bit allocation unit 14 performs a bit allocation process and a quantization process on the sub-band data sb0 to sb31 input from the sub-band division unit 12, based on the bit allocation information from the bit allocation information generation unit 13, Generate quantized data. Thereafter, the multiplexing unit 15 forms the quantized data from the bit allocation unit 14 into an MPEG audio bit stream and outputs the MPEG audio bit stream. The detailed steps of creating bit allocation information in the bit allocation information generation unit 13 and the detailed steps of bit allocation processing and quantization processing using human auditory characteristics in the bit allocation unit 14 are the same as those in the conventional example. Since they are the same as those described above, their duplicate description will be omitted.

As described above, in the digital audio encoding apparatus according to the present embodiment, the sub-band division unit 12 uses the up-sampling information N to perform all of the bands (sub-bands) not included in the input PCM data. The sample is set to zero. As a result, in the digital audio encoding apparatus according to the present embodiment, the filter processing unit in the conventional example shown in FIG. 3 is omitted, and the configuration of the apparatus can be simplified. Furthermore, in the digital audio encoding device of the present embodiment, encoding processing can be performed at high speed without aliasing noise.

In the above description, a one-channel PC
Although the configuration in which the M data is encoded by the MPEG audio encoding unit and formed into an MPEG audio bit stream has been described, the embodiment is not limited to the one-channel PCM data. For example, the MPEG2 audio standard (ISO / IEC 13818- MPEG specified in 3)
It is also possible to generate a multi-channel audio signal from a multi-channel audio signal. further,
The present invention is not limited to the MPEG audio standard. For digital audio signals of the subband encoding system, the encoding process specified by other high-efficiency compression methods can be performed at high speed and with a simple configuration. It is possible to realize with. Further, since any of the encoding methods of the digital audio encoding device in the above-described embodiments can be computer-programmed, it is also possible to provide the encoding method of the present application on a computer-executable recording medium. It is. Here, the recording medium is a data recording device including a floppy disk, a CD-ROM, a DVD, a magneto-optical disk, a removable hard disk, and a flash memory.

[0033]

As described above, according to the digital audio encoding apparatus and the encoding method of the present invention, the sub-band division unit uses the up-sampling information to use the band not included in the input digital audio data. All data of (subband) is set to zero.
Thus, in the digital audio encoding device and the encoding method of the present invention, the configuration of the device can be simplified without providing a filter processing unit.
Further, with the digital audio encoding device and the encoding method of the present invention, encoding processing can be performed at high speed without generating aliasing noise.

In the digital audio encoding apparatus and the encoding method of the invention according to another aspect, when the number of data of one frame of input digital audio data is M (M is a natural number), the upsampling unit
The first to Mth input digital audio data are used to generate first to (M × N−N) th audio data, and the Mth input digital audio data is generated as (M × N−N + 1). ) To (M
× N) th audio data. Thus, in the digital audio encoding device and the encoding method according to the present invention, the processing in the upsampling unit can be reduced, and the encoding processing of the digital audio signal can be performed at high speed. Further, since the encoding method of the present invention can be made into a computer program, the encoding method of the present invention can be recorded on a computer-executable recording medium and executed.

[Brief description of the drawings]

FIG. 1 is a block diagram showing a configuration of a main part of a digital audio encoding device according to an embodiment of the present invention.

FIG. 2 is an explanatory diagram showing an interpolation process in an up-sampling unit shown in FIG. 1;

FIG. 3 is a block diagram showing a configuration of a main part of a conventional digital audio encoding device.

FIG. 4 is an explanatory diagram showing an interpolation process in the upsampling unit shown in FIG. 3;

[Explanation of symbols]

 DESCRIPTION OF SYMBOLS 11 Upsampling part 12 Subband division part 13 Bit allocation information generation part 14 Bit allocation part 15 Multiplexing part

Claims (9)

[Claims] 1. An input digital audio data sampled at a predetermined sampling frequency is up-sampled based on up-sampling information N (N is a factor of 2), and audio data sampled at N times the sampling frequency And a band division process for dividing the audio data from the up-sampling unit into predetermined sub-bands. Further, the sampling frequency is 1 / (2 × N) or more of N times the sampling frequency after the up-sampling. A digital audio encoding device, comprising: a subband division unit that sets all audio data of a subband belonging to a frequency domain to a value of zero. 2. When the number of data for one frame of the input digital audio data is M (M is a natural number), the up-sampling unit uses the first to M-th input digital audio data, The first to (M × N−N) th audio data is generated, and the Mth input digital audio data is used as the (M × N−N + 1) th to (M × N) th audio data. 2. The digital audio encoding device according to claim 1, wherein the digital audio encoding device is configured as described above. 3. The digital audio encoding apparatus according to claim 1, wherein the sub-band dividing unit is configured to divide input audio data into bands based on the MPEG standard. . 4. Up-sampling of input digital audio data sampled at a predetermined sampling frequency based on up-sampling information N (N is a power of 2), and audio data sampled at N times the sampling frequency , And a band division process of dividing the audio data generated in the up-sampling step into predetermined sub-bands, and more than 1 / (2 × N) of N times the sampling frequency after the up-sampling A sub-band division step of setting all audio data of a sub-band belonging to the frequency domain to a value of zero. 5. When the number of data for one frame of the input digital audio data is M (M is a natural number), the first to M-th input digital audio data are used in the upsampling step. The first to (M × N−N) th audio data is generated, and the Mth input digital audio data is converted from the (M × N−N + 1) th to (M × N).
The encoding method of the digital audio encoding device according to claim 4, wherein the audio data is the first audio data. 6. In the sub-band division step,
6. The input audio data is band-divided based on the MPEG standard.
3. The encoding method of the digital audio encoding device according to item 1. 7. Up-sampling of input digital audio data sampled at a predetermined sampling frequency based on up-sampling information N (N is a power of 2), and audio data sampled at N times the sampling frequency , And a band division process of dividing the audio data generated in the up-sampling step into predetermined sub-bands, and more than 1 / (2 × N) of N times the sampling frequency after the up-sampling A sub-band division step of setting all audio data of a sub-band belonging to the frequency domain to a value of zero. 8. When the number of data for one frame of the input digital audio data is M (M is a natural number), in the upsampling step, the first to M-th input digital audio data are used. The first to (M × N−N) th audio data is generated, and the Mth input digital audio data is converted from the (M × N−N + 1) th to (M × N).
8. A recording medium on which the encoding program according to claim 7 is recorded, wherein the encoding data is the first audio data. 9. In the sub-band dividing step,
9. The input audio data is divided into bands based on the MPEG standard.
A recording medium on which the encoding program described in 1 is recorded.