JP2001237708A - Data processing system - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide a data processing system which can prevent the deteriora tion of information. SOLUTION: A level shift processing part 20 bit-shifts inputted 24 bit audio data to a low order by two bits and outputs 26 bit digital data. Digital data of 26 bits, which are bit-shifted, are equalized through an equalization processing part 22 and subjected to an over-sampling processing for four times performed by an over-sampling processing part 24. Then, the resultant data are outputted to a ΣΔ modulation part 26. The ΣΔ modulation part 26 performs a ΣΔ modulation processing on digital data, which are over-sampled to generate 24 bit digital data of 24.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a data processing system for inputting and outputting data of a predetermined number of bits between a plurality of devices in an audio system or the like.

[0002]

2. Description of the Related Art In some audio systems, digital audio data output from a CD player or the like is subjected to various correction processes using a DSP (Digital Signal Processor), and the corrected data is subjected to digital processing. An oversampling process using a filter is performed. In such an audio system, the DSP
When performing a correction process using, for example, a process of raising the signal level of a predetermined band, if the number of bits of the input digital data is maintained, the digital data corresponding to the audio sound having a large signal level is maintained. Is input, the data obtained in the correction processing may exceed the maximum value represented by the number of bits, causing distortion. Therefore, in practice, the signal level corresponding to the input digital data is lowered in consideration of the amount of amplification of the signal generated by the correction processing, and a predetermined correction processing is performed after securing a margin on the upper bit side. I have.

[0003]

As described above, the DSP that performs the predetermined correction processing in the audio system handles data having a larger number of bits than input / output data in the internal processing, and thus outputs data. At this time (when transferring to a digital filter at the subsequent stage), it is necessary to reduce the number of bits that have been increased for performing internal processing, and the loss of information corresponding to this number of bits causes deterioration of information, that is, sound quality. However, there is a problem that the deterioration occurs.

FIG. 7 is a diagram for explaining a correction process performed in the DSP. FIG. 8 is a diagram for explaining the number of bits of data before and after this correction processing. For example, assume that the number of bits of digital data input / output to / from each of the DSP and the digital filter is 16. It is assumed that the DSP performs an equalizer process for emphasizing (amplifying) some frequency bands as shown in FIG. As described above, when the signal level of a part of the frequency band is amplified by the DSP, when 16-bit digital data is input, only a predetermined bit (for example, 2 bits) corresponding to the amplified amount of the entire input data is used. After the bit shift is performed, a predetermined equalizer process is performed. As a result, the number of bits of digital data handled inside the DSP is 18 bits, which is 2 bits added to 16 bits. However, the number of bits of data input / output between the DSP and the digital filter connected to the subsequent stage is one.
6, the DSP sends the internal data 1
16-bit data in which the lower 2 bits of 8 bits are missing is output. Information is deteriorated by the two lower bits that are missing.

[0005] The present invention has been made in view of the above points, and an object of the present invention is to provide a data processing method capable of preventing deterioration of information.

[0006]

In order to solve the above-mentioned problems, according to the data processing method of the present invention, after the input L-bit data is converted into M-bit data by the data conversion means, the data processing means After performing predetermined data processing on the resulting data and performing oversampling processing of a predetermined multiple by the oversampling means on the resulting data, the bit compression means
Performs compression processing to bits. When reducing the number of bits of the data processed, the information of the reduced number of bits is temporally distributed to a plurality of data obtained by the oversampling processing, thereby reducing the number of bits. Can be prevented from deteriorating information.

Therefore, the above-mentioned oversampling processing means performs at least two
It is desirable to perform ^MN times oversampling processing. This makes it possible to reliably disperse the information for the decreasing number of bits into a plurality of data obtained by the oversampling process, and to reliably prevent the information from deteriorating.

The bit compression means described above
It is desirable to obtain N-bit compressed data by performing the Δ modulation process. By combining the ΣΔ modulation processing and the oversampling processing, the number of data bits can be reduced without deteriorating information.

It is desirable to obtain N-bit data excluding the lower-order MN bits by performing dither addition processing by the above-described bit compression means. Since the missing M-N bits of information can be stochastically distributed to a plurality of data by the dither addition processing, it is possible to prevent information deterioration caused by reducing the number of bits. In particular, when the L-bit data input to the above-mentioned data conversion means is audio sound data, the bit compression means makes the data approximately 20 kHz.
It is desirable to add high band concentrated dither band-limited to z or more. By adding the high-frequency concentrated dither band-limited to about 20 kHz or more, it is possible to eliminate the influence on the human hearing when the number of data bits is reduced.

[0010]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS An audio system according to an embodiment to which the data processing system of the present invention is applied will be described below with reference to the drawings.

FIG. 1 is a diagram showing a configuration of an audio system according to the present embodiment. An audio system 100 shown in FIG. 1 includes an audio device 1, a sound quality correction unit 2, an oversampling processing unit 3, a digital-analog (D / A)
Converter 4, low-pass filter (LPF) 5, amplifier 6,
It is configured to include the speaker 7. In FIG. 1, monaural audio data is output from the audio device 1 for the sake of simplicity. However, when stereo audio data is output, May be provided for two or more channels.

The audio apparatus 1 is an audio source that outputs audio data corresponding to various audio signals, and corresponds to a CD player, for example. In the following description, it is assumed that the audio device 1 of the present embodiment outputs 24-bit audio data.

The sound quality correction section 2 performs predetermined data processing on audio data output from the audio device 1 to change the frequency characteristics and the like, thereby performing sound quality correction for emphasizing the high frequency range and the low frequency range. This sound quality correction unit 2
Can be realized by, for example, a DSP.

The oversampling processing section 3 performs oversampling processing on the corrected audio data output from the sound quality correction section 2. This oversampling processing unit 3 can be realized by a digital filter.

The D / A converter 4 converts the audio data output from the oversampling processor 3 into an analog signal. The LPF 5 removes a high-frequency component (for example, a band of 20 kHz or more) of the analog signal output from the D / A converter 4 to change the D-step changing stepwise.
The output signal of the / A converter 4 is smoothed.

The amplifier 6 amplifies the analog signal output from the LPF 5 with a predetermined gain and outputs a signal for driving a speaker. The speaker 7 outputs an audio sound corresponding to the driving signal output from the amplifier 6.

FIG. 2 is a diagram showing a detailed configuration of the sound quality correction section 2. As shown in FIG. 2, the sound quality correction unit 2 includes a level shift processing unit 20, an equalizer processing unit 22, an oversampling processing unit 24, and a ΣΔ modulation unit 26.

The level shift processing section 20 has a function of providing audio data input from the audio device 1 in order to secure a margin for the amplification of signal levels (carry of bits) generated by the equalizer processing by the equalizer processing section 22 at the subsequent stage. Is shifted by a predetermined number of bits to lower the signal level. In this embodiment,
When 24-bit audio data is input, the level shift processing unit 20 performs bit shift to the lower side by 2 bits corresponding to the amplification, and outputs 26-bit digital data. As a result, in the 26-bit data, the upper 2 bits are “0”, and the lower 24 bits include the input audio data.

The equalizer processing section 22 performs equalization processing such as emphasizing (amplifying) a part of the frequency band on the 26-bit digital data output from the level shift processing section 20. For example, as shown in FIG.
Equalizer processing is performed to amplify the signal level in a part of the middle sound range and maintain the signal level in other frequency regions.

The oversampling processing section 24 performs oversampling processing (upsampling processing) on the digital data output from the equalizer processing section 22 after the equalization processing, in which the sampling frequency is increased in a pseudo manner. How many times the oversampling process is performed is determined in consideration of how many bits of the 26-bit digital data are suppressed (compressed) by the ΣΔ modulator 26 at the subsequent stage. For example, when compressing data of M bits (M = 26 in the above example) to N bits,
It is desirable to perform at least 2 ^MN times oversampling processing.

FIG. 3 shows an oversampling processor 24.
FIG. 3 is a diagram showing a specific example of oversampling processing performed by the above method, for example, a case where quadruple oversampling processing is performed. First, the equalizer processing unit 2
2 with respect to the original data input from FIG.
Data (data with a value of 0) is inserted (FIG. 3
(B)). In the case of the quadruple oversampling process, three 0 data are inserted between two adjacent original data. In general, when performing an X-times oversampling process, (X-1) 0 data may be inserted between two adjacent original data.

Thereafter, the data in which one original data and three 0 data are alternately arranged is passed through a low-pass filter in this manner, so that the space between the two adjacent original data is smooth. Three pieces of interpolation data are inserted so as to be connected, and a four-fold oversampling process is performed (FIG. 3C).

The ΣΔ modulation section 26 performs ΣΔ modulation processing on the digital data after the oversampling processing input from the oversampling processing section 24. This Σ
The Δ modulation is a method of converting data amplitude information into a temporal pulse width. By performing the ΔΔ modulation, the number of bits of digital data can be reduced. In the present embodiment, the ΣΔ modulator 26 converts the 26-bit data output from the oversampling processor 24 into 24-bit data equal to the number of bits of the audio data output from the audio device 1.

FIG. 4 is a diagram showing a detailed configuration of the ΣΔ modulator 26. As shown in FIG. 4, the ΣΔ modulator 26 includes two adders 40 and 42, a subtractor 44, two multipliers 46,
48, two delay units 50 and 52, and a quantizer 54.

The adders 40 and 42 add a quantization error delayed by a predetermined delay time to the input digital data. This quantization error is generated when the 26-bit digital data that has passed through the adders 40 and 42 is converted into 24-bit digital data by the quantizer 54, and is input to the quantizer 54. The difference between the digital data before being processed and the digital data output from the quantizer 54 is calculated by the subtractor 44.

The delay unit 52 delays the quantization error output from the subtractor 44 by a time corresponding to one sampling time of the digital data input to the ΣΔ modulation unit 26. Similarly, the delay unit 50 further delays the quantization error output from the delay unit 52 by a time corresponding to one sampling time. The multiplier 48 multiplies the quantization error output from the delay unit 52 by a multiplier “2”. Similarly, the multiplier 46 multiplies the quantization error output from the delay unit 50 by a multiplier “−1”.

The level shift processing section 20 corresponds to data conversion means, the equalizer processing section 22 corresponds to data processing means, the oversampling processing section 24 corresponds to oversampling processing means, and the ΣΔ modulation section 26 corresponds to bit compression means. .

As described above, in the sound quality correction unit 2 included in the audio system 100 of the present embodiment, the input 2
A 2-bit bit shift is performed on the 4-bit audio data to secure a margin on the upper bit side, and then an equalizer process is performed. The resulting 26-bit digital data is subjected to a 4-fold oversampling process. Then, ΣΔ modulation processing is performed to obtain 24-bit digital data. By combining the oversampling process and the ΣΔ modulation process, information for two bits that is reduced when the number of bits of 26-bit digital data after the equalizer process is reduced by two bits can be obtained by the oversampling process. Since a plurality of digital data can be temporally dispersed, it is possible to prevent information deterioration caused by a decrease in the number of bits.

The present invention is not limited to the above embodiment, and various modifications can be made within the scope of the present invention. For example, in the embodiment described above,
The sound quality correction unit 2 compresses 26-bit digital data to 24 bits by performing ΣΔ modulation processing on the digital data after the oversampling processing is performed. The compression processing may be performed by dither addition processing.

FIG. 5 is a diagram showing the configuration of the sound quality correction unit 2a in the case where the number of bits is compressed by dither addition. The sound quality correction unit 2a shown in FIG. 2 has a configuration in which the ΣΔ modulation unit 26 included in the sound quality correction unit 2 shown in FIG. The dither addition unit 28 corresponds to a bit compression unit. Note that each of the level shift processing unit 20, the equalizer processing unit 22, and the oversampling processing unit 24 included in the sound quality correction unit 2a is basically the same as that included in the sound quality correction unit 2 illustrated in FIG. Corresponding components have the same reference characters allotted, and detailed description of operation will be omitted.

The dither addition section 28 performs a predetermined dither addition process on the oversampled 26-bit digital data output from the oversampling processing section 24 to convert the data into 24-bit digital data. In the present embodiment, the dither addition unit 28 performs a process of adding a high-frequency concentrated dither whose band is limited to 20 kHz or more to the input digital data. In this way, by performing the process of adding the high-frequency concentrated dither,
Since the missing two bits of information can be stochastically distributed to a plurality of digital data (digital data obtained by oversampling), it is possible to prevent information deterioration. Further, the high-frequency concentrated dither used in the present embodiment is noise of small amplitude uniformly distributed in a band of 20 kHz or more, so that the audio sound reproduced based on the digital data output from the dither addition unit 28 is reproduced. When listening, there is no influence such as discomfort on the sense of hearing. It should be noted that the low-pass filter having a cut-off frequency of about 20 kHz may be passed in the subsequent stage to remove the minute amplitude caused by adding the high-frequency concentrated dither.

FIG. 6 is a diagram for explaining the principle of the compression process of the number of bits by the dither addition process. FIG. 6A shows the relationship between the input signal in the case where the high frequency concentrated dither is not added and the quantized output corresponding thereto. FIG. 6B shows the relationship between the input signal and the quantized output when the high-frequency concentrated dither is added, respectively.

Simply deleting lower bits without adding high-frequency concentrated dither means that four types of information represented by lower two bits, "00", "01", "10", and "11". Will be missing. As shown in FIG.
Considering the case where the input data is between 1 × Δ and 2 × Δ (Δ = 2 ² because 2 bits are compressed in the present embodiment), for example, these are all quantized as 3/2 values. Output.

However, the dither adding section 28 of the present embodiment
In, a process of adding a high-frequency concentrated dither of ± Δ / 2 (= ± 2) to input data and then deleting lower two bits is performed. For example, as shown in FIG. 6B, when the input data is (1 + 3/4) × Δ, by adding a dither of ± Δ / 2 to this input data, 1.25Δ to Data with a distribution having a width of 2.25Δ is obtained. In practice, since one data included in this distribution is obtained, 24
Performing bit quantization means that this data is 1.
The quantization output becomes 3/2 when included in the range of 75% corresponding to 25Δ to 2Δ, and becomes quantized output when included in the range of 25% corresponding to 2Δ to 2.25Δ. Is 5/2. Therefore, the average over time is 1.75.

As described above, by performing dither addition processing instead of performing ΣΔ modulation processing, information of two missing bits is temporally dispersed, thereby preventing deterioration of information caused by a decrease in the number of bits. can do.

In the above-described embodiment, the case where the number of bits of the output data of the sound quality correction unit constituted by the DSP or the like is compressed is described. However, the output data of the DSP or another device used for other purposes is used. The present invention can be applied to the case where the number of bits is compressed.

In the description of the above-described embodiment, the number of bits of data input to and output from the sound quality correction unit is both 24 bits. However, when the number of bits of input data is different from the number of bits of output data, It may be.

[0038]

As described above, according to the present invention, the information corresponding to the reduced number of bits is temporally distributed to a plurality of data obtained by the oversampling process, whereby the number of bits is reduced. It is possible to prevent information deterioration caused by the decrease.

[Brief description of the drawings]

FIG. 1 is a diagram showing a configuration of an audio system.

FIG. 2 is a diagram illustrating a detailed configuration of a sound quality correction unit.

FIG. 3 is a diagram illustrating a specific example of oversampling processing performed by an oversampling processing unit.

FIG. 4 is a diagram illustrating a detailed configuration of a ΣΔ modulation unit.

FIG. 5 is a diagram illustrating a configuration of a sound quality correction unit when performing compression processing of the number of bits by dither addition processing.

FIG. 6 is a diagram for explaining the principle of compression processing of the number of bits by dither addition processing.

FIG. 7 is a diagram illustrating a correction process performed in a DSP in a conventional audio system.

FIG. 8 is a diagram illustrating the number of bits of data before and after a correction process.

[Explanation of symbols]

 Reference Signs List 1 audio device 2, 2a sound quality correction unit 3 oversampling processing unit 4 digital-analog (D / A) converter 5 low pass filter (LPF) 6 amplifier 7 speaker 20 level shift processing unit 22 equalizer processing unit 24 oversampling processing unit 26 ΣΔ modulator 28 dither adder

Claims (5)

[Claims] 1. A data conversion means for converting input L-bit data into M-bit data longer than L-bit data, and a predetermined data processing for the M-bit data output from the data conversion means. A data processing unit that performs oversampling processing on data output from the data processing unit; a bit number compression process on M-bit data output from the oversampling processing unit And a bit compression means for converting the data into N-bit data shorter than M bits. 2. The data processing method according to claim 1, wherein said oversampling processing means performs at least 2 ^MN times oversampling processing on input data. 3. The data processing method according to claim 1, wherein said bit compression means obtains N-bit compressed data by performing ΣΔ modulation processing. 4. The bit compression device according to claim 1, wherein the bit compression unit performs dither addition processing.
A data processing method for obtaining N-bit data excluding lower-order MN bits. 5. The L-bit data input to the data conversion means according to claim 4,
A data processing method for audio sound data, wherein the bit compression means adds a high-frequency concentrated dither whose band is limited to approximately 20 kHz or more.