KR101471978B1 - Method for inserting data for enhancing quality of audio signal and apparatus therefor - Google Patents

Abstract

The present invention relates to a method of inserting data for improving the quality of an audio signal. The method includes generating a first signal by normalizing an audio signal, downsampling and quantizing the first signal to generate a second signal, Sampling the quantized signal to generate a third signal by performing up-sampling and bit-shifting, calculating an error by differentiating the third signal from the first signal, encoding the calculated error, and inserting the error into the second signal, It is possible to effectively improve the sound quality of the audio signal while minimizing the noise generated in the audio signal.

Beat, sound quality, enhance, reinforce, insert

Description

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a method for inserting data for enhancing audio quality of an audio signal,

FIG. 1 is a view for explaining an embodiment of a data inserting apparatus for improving sound quality of a conventional audio signal.

2 is a block diagram of an apparatus for inserting data for enhancing sound quality of an audio signal according to an embodiment of the present invention.

3 is a diagram illustrating an embodiment of error data according to the present invention.

4 is a diagram illustrating an embodiment of an error data packet to be inserted into an audio signal according to the present invention.

5A and 5B are diagrams for explaining a pattern of the number of bits used for insertion of error data according to the present invention.

FIG. 6 is a flowchart illustrating a data insertion method for enhancing sound quality of an audio signal according to the present invention.

7 is a diagram illustrating an apparatus for enhancing audio quality of an audio signal using embedded data according to an embodiment of the present invention.

FIG. 8 is a flowchart illustrating a method of improving sound quality of an audio signal using embedded data according to the present invention.

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a data insertion method and apparatus for improving the sound quality of an audio signal, a method for improving the sound quality of an audio signal using the inserted data, and an apparatus therefor.

The PCM (Pulse Code Modulation) method is a method of sampling analog data at a predetermined cycle, quantizing the data, and performing binary coding on 8, 16, 32, or 64 bits.

 The bit robbing method is a method of using a predetermined bit of a PCM sample to insert a PCM encoded sample signal, that is, information irrelevant to the information possessed by the original PCM sample, into the PCM sample. For example, to carry a given content, it uses regular bits of the PCM sampled data regularly to form a data channel independent of the PCM data.

Such a bit-rob scheme is used in a public switched telephone network (PSTN) and in a T-carrier system widely used for voice and data transmission in a private network. In such a system, information about which bits of the PCM samples will be used should be known in advance. Such a method is disclosed in U.S. Patent No. 5,864,600.

FIG. 1 is a view for explaining an embodiment of a data inserting apparatus for improving sound quality of a conventional audio signal.

Referring to FIG. 1, a data insertion unit for improving the sound quality of a conventional audio signal includes an AD converter 110, a normalization unit 120, a downsampling unit 130, a quantization unit 140, and a data insertion unit 150. .

The AD converter 110 converts an analog audio signal into a digital audio signal, and the normalization unit 120 normalizes the audio signal. Normalization is also referred to as digital mastering, and refers to matching the volume of audio signals stored in an audio CD or the like to a constant level.

The downsampling unit 130 lowers the sampling rate of the normalized audio signal.

The quantization unit 140 re-quantizes the number of bits of the audio signal whose sampling rate is lowered.

Since the sampling rate and the number of bits of an analog signal are very high, it is necessary to lower the sampling rate and the number of bits in order to record on a CD or the like. In the downsampling unit 130 and the quantization unit 140, Function. Audio CDs typically have a sampling rate of 44.1 KHz and audio signals recorded at 16 bits.

The data inserting unit 150 inserts data, which can improve the sound quality of the audio signal, into the audio signal. At this time, data that can improve the sound quality of the audio signal include information for selecting gain, peak limiting, and decimation filter. Here, the decimation is a technique used to reduce the amount of sampled data, and the information for decimation filter selection is information for selecting such a filter.

By inserting such data, an audio signal having a sampling rate of, for example, 16 bits / 44.1 KHz can be converted into an audio signal having a sampling rate of 20 bits / 192 KHz.

However, the conventional method of improving the sound quality of an audio signal does not effectively improve the sound quality, and further, there is a problem that the sound quality of the audio signal is lowered as the data is inserted.

It is an object of the present invention to provide a data inserting method and apparatus for enhancing sound quality of an audio signal for effectively improving the sound quality of an audio signal while minimizing noise generated in the audio signal, Device.

According to another aspect of the present invention, there is provided a method of inserting data for enhancing audio quality of an audio signal, the method comprising: generating a first signal by normalizing the audio signal; Performing downsampling and quantization on the first signal to generate a second signal; Performing upsampling and bit shifting on the quantized signal to generate a third signal; Calculating error data by differentiating the third signal from the first signal; And encoding and encoding the calculated error data into the second signal.

Preferably, the inserting step inserts the error data into a predetermined bit of the audio signal sample based on the energy level of the audio signal.

According to another aspect of the present invention, there is provided a method for enhancing audio quality of an audio signal using embedded data, comprising: extracting error data inserted in a predetermined bit of the audio signal sample; Generating a first signal by decoding the extracted error data, generating a second signal by performing up-sampling and bit-shifting on an audio signal of a bit other than a predetermined number of bits into which the error is inserted; And combining the first signal and the second signal.

According to another aspect of the present invention, there is provided a data insertion apparatus for enhancing audio quality of an audio signal, comprising: a first signal generator for generating a first signal by normalizing the audio signal; A second signal generator for generating a second signal by performing downsampling and quantization on the first signal; A third signal generator for performing up-sampling and bit-shifting on the quantized signal to generate a third signal; An error calculator for calculating error data by differentiating the third signal from the first signal; And an error insertion unit for encoding the error data and inserting the encoded error data into the second signal.

According to another aspect of the present invention, there is provided an apparatus for enhancing audio quality of an audio signal using embedded data, comprising: an error extracting unit for extracting error data inserted in a predetermined bit of the audio signal sample; A first signal generator for decoding the extracted error data to generate a first signal; A second signal generator for generating a second signal by performing up-sampling and bit-shifting on an audio signal of a bit other than a predetermined bit into which the error data is inserted; And a signal combining unit for combining the first signal and the second signal.

Hereinafter, preferred embodiments of the present invention will be described with reference to the accompanying drawings.

2 is a block diagram of an apparatus for inserting data for enhancing sound quality of an audio signal according to an embodiment of the present invention.

2, the apparatus for inserting data according to the present invention includes a first signal generator 210, a second signal generator 220, a third signal generator 230, an error calculator 240, (250).

The first signal generator 210 normalizes the audio signal.

Hereinafter, a signal obtained by normalizing the audio signal in this way is referred to as a first signal.

The second signal generation unit 220 includes a downsampling unit 222 and a quantization unit 224. [

The downsampling unit 222 lowers the sampling rate of the first signal.

The quantization unit 224 lowers the number of bits of the downsampled first signal.

Hereinafter, a signal obtained by lowering the sampling rate and the bit number of the first signal is called a second signal.

The reason why the second signal generator 220 is processed in this way is that the sampling rate and the number of bits of the audio signal input to the first signal generator 210 are very high. To be reduced to the number of bits.

The third signal generating unit 230 includes an upsampling unit 232 and a bit shifting unit 234. [

The upsampling unit 232 increases the sampling rate of the second signal.

The bit shifting unit 234 increases the number of bits of the second signal having a higher sampling rate.

The reason why the signal is processed by the third signal generator 230 is that the error signal of the first signal and the signal of the third signal is sampled by the error calculator 240, This is because the difference between the rate and the number of bits is only possible.

That is, the third signal generator 230 increases the sampling rate and the number of bits of the second signal, and generates a signal that matches the first signal with the sampling rate and the number of bits.

The error calculator 240 calculates error data by differentiating the third signal from the first signal.

In detail, the error calculator 240 subtracts a signal obtained by performing down-sampling, quantization, up-sampling, and bit-shifting on a signal obtained by normalizing the input audio signal (first signal) It calculates the data.

3 is a diagram illustrating an embodiment of error data according to the present invention.

3 is a diagram showing an error in an audio signal having a sampling rate of 44.1 kHz. First, a quantization error is shown at 22.05 KHz or less. This is an error occurring in the process of quantizing by lowering the audio signal to a low number of bits such as 16 bits for recording an audio signal in an audio CD or the like in the second signal generator 220.

In addition, the graph waveform of the input audio signal is shown as an error at 22.05 KHz or more. In order to reproduce the audio signal without damaging it, it is an error that occurs due to the restriction that sampling is performed at a frequency which is twice or more the maximum frequency of the audio signal when sampling.

That is, if the audio signal is sampled at a sampling rate of 44.1 KHz, a signal of 22.05 KHz or more is forcibly removed due to the above limitation. Therefore, the audio signal of 22.05 KHz or higher is not present in the second signal. Therefore, if the third signal which is a signal that increases the sampling rate and the bit rate of the second signal in the first signal is subtracted, the audio signal of 22.05 KHz in the first signal is obtained as an error at 22.05 KHz or more.

The error insertion unit 250 encodes the error data calculated by the error calculation unit 240 and inserts the error data into the second signal. At this time, the error inserting unit 250 inserts the error data calculated in the predetermined bit of the audio signal sample based on the energy level of the audio signal.

At this time, the encoding method can be MP3, MPEG, SBR (Spectral Band Replication).

The reason for inserting the error data calculated on a predetermined bit based on the energy level is to prevent noise from being generated in the audio signal while inserting such error data.

That is, when the energy level of the input audio signal is equal to or greater than the predetermined level, even if data is inserted into a plurality of least significant bits (LSBs) of the audio sample, the resulting noise is not heard by the original audio signal. It is to use this characteristic.

In addition, the error insertion unit 250 may randomly insert error data.

3 is a view illustrating a method of inserting data according to an energy level according to the present invention.

FIG. 3 illustrates simultaneous-masking, in which masking is performed by masking notes, pre-masking before and after masking, and forward masking. As shown in FIG. 3, a masking effect, that is, a backwardness and an omnidirectional masking effect, appears before and after a masking sound, which is called a temporal masking effect.

Here, in the case of simultaneous-masking, the masking effect appears in proportion to the negative pressure of the masking sound. In addition, the instantaneous masking effect becomes larger as the time difference from the masking sound becomes shorter. Hereinafter, such a masking effect is used to insert the error data into the audio signal within a range where the listener can not perceive degradation in sound quality.

In the present invention, an audio signal is divided into predetermined blocks for energy level calculation, and each divided block is called an audio block. As a result of using the masking effect as described above in the present invention, even when the energy level of the current audio block is low as described later, when the energy level of the previous audio block or the audio block thereafter is intermediate or high, .

4 is a diagram illustrating an embodiment of an error data packet to be inserted into an audio signal according to the present invention.

4, an error data packet according to the present invention includes a sync word, duration information, bit robbing information, and the like. pattern information and error data.

The start synchronization word indicates the start of the error data packet. In this embodiment, the start sync word uses 16 bits. The 16 bits are of sufficient length as a start code and the probability of false detection is very low. In the present embodiment, the synchronization word is inserted using the least significant bit of the PCM sample without regard to the energy level.

The duration information indicates the number of PCM samples used to insert the error data. At this time, the period information uses 16 bits following the sync word and is inserted using the least significant bit of the PCM sample. The information including the information on the number of PCM samples into which the error data is inserted in the period information is such that even if the energy level is at an intermediate level or a high level, if the error data to be inserted is already inserted, To indicate that the bit-rob was not performed.

The error data is adaptively inserted into the PCM samples based on the energy levels of the current audio block, the previous audio block, and the subsequent audio block.

For example: i) if the energy level of the current audio block is at an intermediate level, the error data is inserted using one least significant bit of every PCM sample.

ii) Also, if the energy level of the current audio block is at a high level, the error data is inserted using a plurality of least significant bits of every PCM sample.

iii) Furthermore, if the energy level of the current audio block is at a low level and the energy level of the previous or subsequent audio block is medium or high level, then the error data uses the least significant bits of the PCM samples at intervals of predetermined PCM samples . For example, error data can be inserted using the least significant bits of the PCM samples at five PCM sample intervals.

In this embodiment, the number of bits that are bit-worn according to the energy level of the current audio block, the previous audio block, and the following audio block is divided into one or a plurality of bits, but the number of bits that are bit- It can be determined in a different pattern.

5A and 5B are diagrams for explaining a pattern of the number of bits used for insertion of error data according to the present invention.

FIG. 5A shows insertion of error data using one or more bits for every PCM sample. When the energy level of the current audio block is at a medium level or a high level, error data is inserted in such a pattern.

FIG. 5 (b) shows an insertion using a least significant bit for every predetermined number of PCM samples. When the energy level of the current audio block is low and the energy level of the previous audio block or the subsequent audio block is medium or high level Error data is inserted in this pattern.

The bit pattern information includes information on the number of bits used for error data insertion in the PCM samples. Optionally, the bit-rob pattern information may be used to indicate that the position information of the bits used for error data insertion, e. G. Error data, is inserted at five PCM sample intervals.

The end sync word indicates that all error data packets have been inserted. In this embodiment, the length of the termination sync word is 16 bits.

FIG. 6 is a flowchart illustrating a data insertion method for enhancing sound quality of an audio signal according to the present invention.

In operation 610, the audio signal is normalized to generate a first signal.

In operation 620, downsampling and quantization are performed on the first signal to generate a second signal.

In operation 630, upsampling and bit shifting are performed on the quantized signal to generate a third signal.

In step 640, the error signal is calculated by subtracting the third signal from the first signal.

In step 650, the calculated error data is encoded and inserted into the second signal.

As described above, at this time, the error insertion unit 250 inserts the error data calculated in predetermined bits of the audio signal samples based on the energy level of the audio signal, minimizes the noise generated in the audio signal, can do.

7 is a diagram illustrating an apparatus for enhancing audio quality of an audio signal using embedded data according to an embodiment of the present invention.

7, the sound quality enhancing apparatus according to the present invention includes an error extracting unit 710, a first signal generating unit 720, a second signal generating unit 730, and a signal combining unit 740.

The error extracting unit 710 extracts error data embedded in a predetermined bit of the audio signal sample.

The operation of the error extracting unit 710 will be described in more detail. First, when the error data is randomized and inserted, the error extracting unit 710 descrambles the inserted data And then extracts the start sync word. However, the descrambling process may be omitted unless the error data is randomly inserted.

When the start synchronizing word is valid, the error extracting unit 710 extracts the period information and the bit pattern information shown in FIG. 4, and then, based on the extracted period information, the bit pattern information and the energy level of the audio signal, Data is extracted.

For example: i) if the energy level of the current audio block is at an intermediate level, the error data is extracted from one least significant bit of every PCM sample.

ii) Also, if the energy level of the current audio block is at a high level, the error data is extracted from a plurality of multiple bits of every PCM sample.

Finally, if the energy level of the current audio block is at a low level and the energy level of the previous or later audio block is at a medium or high level, then the error data is calculated from the least significant bits of the PCM samples at intervals of predetermined PCM samples And extracted. For example, error data can be extracted from the least significant bits of the PCM samples at five PCM sample intervals.

The first signal generator 720 decodes the error data extracted from the error extractor 710. Hereinafter, a signal obtained by decoding the extracted error data will be referred to as a first signal. At this time, the first signal generator 720 performs decoding in a manner corresponding to the manner in which the error data is encoded. That is, when the encoding is performed by a method such as MP3, MPEG, or SBR (Spectral Band Replication), decoding is performed in a manner corresponding thereto.

The second signal generator 730 performs upsampling and bit-shifting on the audio signal recorded in bits other than the bit in which the error is inserted in the audio signal.

The audio signal on which the upsampling and bit shifting are performed is hereinafter referred to as a second signal.

The upsampling unit 732 increases the sampling rate of the audio signal recorded in bits other than the bit where the error is inserted in the audio signal.

The bit shifting 734 increases the number of bits of the audio signal with a high sampling rate.

The reason for increasing the sampling rate and the number of bits is that the first signal and the second signal are combined in the signal combining unit 740 to be described later. In this case, the sampling rate of the first signal and the second signal, The number should be the same.

For example, if the sampling rate and the number of bits of the first signal are 192 KHz and 20 bits, respectively, and the sampling rate and the number of bits of the second signal are 44.1 KHz and 16 bits, Give.

The signal combiner 740 combines the first signal and the second signal.

More specifically, the signal combiner 740 combines a signal (first signal) obtained by extracting and decoding error data from an audio signal and a signal (second signal) obtained by performing up-sampling and bit-shifting on an audio signal other than the error data, The sound quality of the audio signal is improved.

FIG. 8 is a flowchart illustrating a method of improving sound quality of an audio signal using embedded data according to the present invention.

In step 810, error data embedded in a predetermined bit of the audio signal sample is extracted.

In step 820, the extracted error data is decoded to generate a first signal.

In step 830, upsampling and bit shifting are performed on the audio signal of the bits other than the predetermined bit into which the error data is inserted to generate the second signal.

In step 840, the first signal and the second signal are combined.

The above-described embodiments of the present invention can be embodied in a general-purpose digital computer that can be embodied as a program that can be executed by a computer and operates the program using a computer-readable recording medium.

The computer readable recording medium may be a magnetic storage medium such as a ROM, a floppy disk, a hard disk, etc., an optical reading medium such as a CD-ROM or a DVD and a carrier wave such as the Internet Lt; / RTI > transmission).

The present invention has been described with reference to the preferred embodiments. It will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the spirit and scope of the invention as defined by the appended claims. Therefore, the disclosed embodiments should be considered in an illustrative rather than a restrictive sense. The scope of the present invention is defined by the appended claims rather than by the foregoing description, and all differences within the scope of equivalents thereof should be construed as being included in the present invention.

The present invention includes a method for generating a first signal by normalizing an audio signal, downsampling and quantizing the first signal to generate a second signal, performing up-sampling and bit-shifting on the quantized signal After generating the third signal, the error is calculated by calculating the error by subtracting the third signal from the first signal, and the calculated error is encoded and inserted into the second signal to minimize the noise generated in the audio signal, There is an effect of improving effectively.

Claims (5)

A data insertion method for improving sound quality of an audio signal, Normalizing the audio signal to produce a first signal; Performing downsampling and quantization on the first signal to generate a second signal; Performing upsampling and bit shifting on the quantized signal to generate a third signal; Calculating error data by differentiating the third signal from the first signal; And And encoding and encoding the calculated error data into the second signal The step of encoding the error data and inserting the calculated error data into the second signal uses a masking effect, The error data is inserted in the form of an error data packet. The error data packet includes information on the number of bits used for inserting the error data among the PCM (Pulse Code Modulation) samples and the error data Lt; / RTI > The method according to claim 1, The inserting step And inserting the error data into a predetermined bit of a sample based on the audio signal based on the energy level of the audio signal. A method for enhancing audio quality of an audio signal using embedded data, Extracting error data inserted into a predetermined bit of a sample based on the audio signal using a masking effect; Decoding the extracted error data to generate a first signal; Performing up-sampling and bit-shifting on an audio signal of a bit other than a predetermined number of bits into which the error data is inserted to generate a second signal; And And combining the first signal and the second signal The error data is inserted in the form of an error data packet. The error data packet includes information on the number of bits used for inserting the error data among PCM (Pulse Code Modulation) samples and the error data A method for enhancing sound quality. A data insertion apparatus for improving sound quality of an audio signal, A first signal generator for generating a first signal by normalizing the audio signal; A second signal generator for generating a second signal by performing downsampling and quantization on the first signal; A third signal generator for performing up-sampling and bit-shifting on the quantized signal to generate a third signal; An error calculator for calculating error data by differentiating the third signal from the first signal; And And an error insertion unit for encoding the calculated error data and inserting the encoded error data into the second signal The error insertion unit uses a masking effect The error data is inserted in the form of an error data packet. The error data packet includes information on the number of bits used for inserting the error data among the PCM (Pulse Code Modulation) samples and the error data . An apparatus for enhancing audio quality of an audio signal using embedded data, An error extracting unit for extracting error data inserted into a predetermined bit of the sample based on the audio signal using a masking effect; A first signal generator for decoding the extracted error data to generate a first signal; A second signal generator for generating a second signal by performing up-sampling and bit-shifting on an audio signal of a bit other than a predetermined number of bits into which the error data is inserted; And And a signal combining unit for combining the first signal and the second signal The error data is inserted in the form of an error data packet. The error data packet includes information on the number of bits used for inserting the error data among the PCM (Pulse Code Modulation) samples and the error data A sound quality improving device.

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