KR20080023618A - Audio encoding and decoding apparatus and method using warped linear prediction coding - Google Patents

Abstract

An apparatus for audio encoding and decoding using warped linear prediction coding, and a method thereof are provided to remove the redundancy of an original signal by using the warped linear prediction coding in an audio encoding process, provide an error signal to an audio encoder as an input signal, and transform a psychological sound model to be suitable for the error signal, thereby increasing the efficiency of audio signal compression and performing audio signal encoding. An error signal calculating unit(110) performs the warped linear prediction coding of an audio signal inputted from the outside in a temporal area to calculate an error signal. A frequency domain converting unit(120) converts the error signal obtained in the error signal calculating unit into a frequency domain signal. A masking threshold value calculating unit(131,132) calculates a masking threshold value used in the encoding of the error signal by using an original signal and encoding information used in the warped linear prediction coding of the original signal. A perceptual encoding unit(140) performs the perceptual encoding of the error signal converted in the frequency domain converting unit by using the calculated masking threshold value.

Description

Audio Encoding and Decoding Apparatus Using Modified Linear Prediction Coding and Method therefor {Audio Encoding and Decoding Apparatus and Method using Warped Linear Prediction Coding}

1 is a configuration diagram of an audio encoding apparatus using modified linear prediction encoding according to the present invention;

FIG. 2 is a diagram for explaining a process of extracting a masking threshold for an error signal from a masking threshold for an original signal in an audio encoding method using modified linear prediction coding according to the present invention;

3 is a diagram illustrating an embodiment of an input frame in a modified discrete cosine transform process and a psychoacoustic model process applied to the present invention;

4 is a configuration diagram of an audio decoding apparatus using modified linear prediction encoding according to the present invention.

* Explanation of symbols on the main parts of the drawing

110: modified linear prediction encoder 120: modified discrete cosine transform unit

130: psychoacoustic model unit 131: first masking threshold calculation unit

132: second masking threshold calculator 140: perceptual encoder

150: bitstream packing unit 410: bitstream extracting unit

420: perceptual decoding unit 430: inverse transform discrete cosine transform unit

440: modified linear prediction decoder

The present invention relates to an audio encoding and decoding apparatus using modified linear predictive encoding, and more particularly, to an error signal obtained by linear predictive encoding of an input audio signal and a masking threshold of the error signal (hereinafter, referred to as an error masking threshold). ), And the linearly predictive decoding of the error signal obtained by decoding the encoded bit stream using linear predictive encoding information (linear prediction analysis coefficient), thereby improving the compression efficiency of audio encoding. The present invention relates to an audio encoding and decoding apparatus using predictive encoding and a method thereof.

The Moving Picture Experts Group (MPEG) of the International Organization for Standardization (ISO) / International Electrotechnical Commission (IEC), a standardization body for audio codecs, is known as the 'MPEG-1 / 2 Layer III', Standards such as MPEG-4 HE-AAC ', and' BSAC 'have been developed. These standards commonly apply a psychoacoustic model (PAM) in the frequency domain to minimize signal redundancy and include entropy coding.

In exceptional cases, the codec technique through prediction determines whether to perform encoding through prediction in consideration of a gain when removing signal redundancy through prediction when encoding a signal of some frames in the time domain. Done.

Meanwhile, in the case of a general audio signal, various sound sources are mixed with each other. Therefore, encoding through prediction in the time domain suitable for a single sound source signal (periodic signal) is not mainly used. Codec technology through prediction in the time domain is used in a limited way when the prediction gain of some specific frames is high. In contrast, the conventional codec method in the frequency domain has a high compaction of signals in the frequency domain, particularly in the low frequency domain, so that a high codec gain can be obtained through the codec method in the frequency domain.

However, the conventional codec technique in the frequency domain has a Fast Fourier Transform (FFT) for transforming into the frequency domain and a spreading function that applies masking characteristics to calculate the masking threshold in the psychoacoustic model. Processing, tonality processing through linear inter-frame prediction, etc. are performed, requiring a lot of calculation processes. In addition to the fast Fourier transform in the psychoacoustic model, transform discrete cosine transform is performed on the time-domain signal for signal processing in the frequency domain. Such a conventional codec technology can perform audio compression at high quality through a psychoacoustic model, but there is a problem in that a complicated processing process and a calculation amount of data are increased.

The present invention has been proposed to solve the above problems, and an error signal obtained by linear predictive encoding of an input audio signal is encoded using a masking threshold of the error signal (hereinafter, referred to as an error masking threshold), and the encoded bit string is encoded. An apparatus and method for audio encoding and decoding using modified linear prediction encoding, by which linear prediction decoding of the error signal obtained by decoding is performed using linear prediction encoding information (linear prediction analysis coefficient), thereby improving the compression efficiency of audio encoding. The purpose is to provide.

Other objects and advantages of the present invention can be understood by the following description, and will be more clearly understood by the embodiments of the present invention. It will also be appreciated that the objects and advantages of the present invention may be realized by the means and combinations thereof indicated in the claims.

According to an aspect of the present invention, there is provided an audio encoding apparatus using modified linear predictive encoding, comprising: error signal calculating means for obtaining an error signal by linearly predicting an audio signal (original signal) input from the outside in a time domain; Frequency domain conversion means for converting the error signal obtained by the error signal calculation means into a frequency domain signal; Masking threshold calculation means for obtaining a masking threshold value used for encoding the error signal using the original signal and encoding information used for linear prediction encoding of the original signal; And perceptual encoding means for perceptually encoding the error signal converted by the frequency domain transforming means using the obtained masking threshold.

On the other hand, the present invention provides an audio decoding apparatus using modified linear prediction encoding, comprising: bitstream extracting means for extracting a bit string corresponding to a current frame from all bitstreams transmitted from the outside; Perceptual decoding means for perceptually decoding the bit stream extracted by the bit stream extracting means to obtain the error signal of the current frame and the encoding information used for the linear prediction encoding; Error signal conversion means for converting the error signal obtained by the decoding means into an error signal in a time domain; And modified linear predictive decoding means for generating an audio signal by reconstructing the error signal converted by the error signal converting means using the encoding information used in the linear predictive encoding obtained by the perceptual decoding means.

On the other hand, the present invention is an audio encoding method using modified linear prediction encoding, comprising: an error signal calculation step of obtaining an error signal by linearly predicting and encoding an audio signal (original signal) input from the outside in a time domain; A frequency domain conversion step of converting the error signal obtained in the error signal calculation step into a frequency domain signal; A masking threshold calculation step of calculating a masking threshold value used for encoding the error signal using the original signal and encoding information used for linear prediction encoding of the original signal; And a perceptual encoding step of perceptually encoding the error signal converted in the frequency domain transforming step using the masking threshold value obtained.

On the other hand, the present invention, in the audio decoding method using the modified linear prediction coding, Bitstream extraction step of extracting a bit string corresponding to the current frame from the entire bitstream transmitted from the outside; A perceptual decoding step of perceptually decoding the bit stream extracted in the bit stream extraction step to obtain the error signal of the current frame and the encoding information used for the linear prediction encoding; An error signal conversion step of converting the error signal obtained in the decoding step into an error signal in a time domain; And a modified linear prediction decoding step of generating an audio signal by reconstructing the error signal converted in the error signal conversion step using the encoding information used in the linear prediction encoding obtained in the perceptual decoding step.

The above objects, features and advantages will become more apparent from the following detailed description taken in conjunction with the accompanying drawings, whereby those skilled in the art may easily implement the technical idea of the present invention. There will be. In addition, in describing the present invention, when it is determined that the detailed description of the known technology related to the present invention may unnecessarily obscure the gist of the present invention, the detailed description thereof will be omitted. Hereinafter, exemplary embodiments of the present invention will be described in detail with reference to the accompanying drawings.

1 is a configuration diagram of an audio encoding apparatus using modified linear prediction encoding according to the present invention.

As shown in FIG. 1, an audio encoding apparatus using modified linear prediction encoding according to the present invention includes a modified linear prediction encoding unit 110, a modified discrete cosine transform unit 120, a psychoacoustic model unit 130, and perceptual The encoder 140 and the bitstream packing unit 150 are included. Here, the psychoacoustic model unit 130 includes a first masking threshold calculator 131 and a second masking threshold calculator 132.

In order to encode a block of a predetermined size in the current frame, the modified linear prediction encoder 110 passes a Warped Linear Prediction Coding filter on the current block to obtain a residual component obtained by subtracting an estimate. Here, the residual component means an error signal of the PCM audio signal compared to the estimated value. That is, the modified linear prediction encoder 110 obtains an error signal by performing a modified linear prediction encoding on a pulse code modulation (PCM) audio signal (original signal) input from the outside in the time domain. The modified linear prediction encoder 110 transmits the error signal to the modified discrete cosine transform unit 120.

The modified linear prediction encoder 110 obtains encoding information (Analysis Coefficients of the modified linear prediction encoding) and transmits the analysis coefficients to the psychoacoustic model unit 130. The analysis coefficients of the modified linear prediction encoding may be analysis coefficients of the synthesis filter of the modified linear prediction encoding. This analysis coefficient is used in the psychoacoustic model unit 130 to normalize the masking threshold.

The modified discrete cosine transform unit 120 performs a modified discrete cosine transform (MDCT) on the error signal transmitted from the modified linear predictive encoding unit 110. This is to convert the error signal in the time domain to the frequency domain.

The first masking threshold calculator 131 performs Fast Fourier Transform (FFT) operation, Partial sum operation, and U.S. operation on the transmitted PCM audio signal to reflect the auditory characteristics of the human frequency domain. A masking threshold (hereinafter, referred to as an original signal masking threshold) for the PCM audio signal is obtained by performing a prediction measure and a spreading process.

The second masking threshold calculator 132 receives an analysis coefficient (encoding information) of the modified linear prediction encoding from the modified linear prediction encoder 110 and an original signal masking threshold from the first masking threshold calculating unit 131. The second masking threshold calculator 132 obtains a masking threshold (hereinafter, referred to as an error masking threshold) for an error signal reflecting the perceptual characteristics of the audio signal using the original signal masking threshold and the analysis coefficients of the modified linear prediction coding. The second masking threshold calculator 132 calculates a signal-to-masking ratio (SMR) (hereinafter, error masking SMR) with respect to the error masking threshold using the error masking threshold. The psychoacoustic model unit 130 transfers the calculated error masking SMR to the perceptual encoder 140.

After that, the calculation of the error masking threshold will be described in detail with reference to FIG. 2.

The perceptual encoder 140 performs perceptual encoding on the error signal using the error masking SMR received from the psychoacoustic model unit 130 in the modified discrete cosine transform region (frequency domain). That is, the perceptual encoder 140 quantizes the error signal using the error masking SMR. The error masking threshold is used as a reference for quantization noise to quantize the error signal.

The bitstream packing unit 150 entropy-codes the error signal transmitted from the perceptual encoding unit 140 and packs it into a bit string.

FIG. 2 is a diagram illustrating an example of a process of extracting an error masking threshold value from a masking threshold value of an original signal in an audio encoding method using modified linear predictive encoding according to the present invention.

The first masking threshold calculator 131 obtains an original signal masking threshold from a partition band in which a perceptual characteristic is reflected without obtaining a masking threshold in a linear band (210). Here, the linear band has a fine frequency band and all bands have the same frequency band spacing, and thus have a linear frequency resolution. In contrast, the partition band has a nonlinear frequency resolution because the frequency band in the low frequency region is finer than the frequency band in the high frequency region. In other words, the partition band has a frequency resolution that reflects a person's perceptual characteristics.

The second masking threshold calculator 132 calculates the original signal masking threshold in the linear band by mapping the original signal masking threshold obtained in the partition band to the linear band again (220). In this case, components in a linear band belonging to an arbitrary partition band have the same value by dividing the threshold of the partition band by the frequency bin number of the linear band. Equation 1 below shows a process of mapping a partition band to a linear band.

는 선형 밴드에서의 원신호 마스킹 임계치,

는 파티션 밴드에서의 원신호 마스킹 임계치,

와

는 각 파티션 밴드에 해당되는 선형 밴드의 주파수 경계값을 나타낸다. here,

Is the original signal masking threshold in the linear band,

Is the original signal masking threshold in the partition band,

Wow

Denotes a frequency boundary value of the linear band corresponding to each partition band.

As shown in Equation 1, the original signal masking threshold in the linear band is obtained. If the original signal masking threshold is, the error masking threshold is normalized to the energy of the analysis coefficients of the modified linear prediction coding as shown in Equation 2 below. That is, when the error masking threshold is normalized by the energy of the transfer function H (z) of the modified linear prediction coding, the error masking threshold in the linear band is obtained. Here, the transfer function H (z) of the modified linear prediction coding corresponds to the analysis coefficient of the modified linear prediction coding.

는 오차 마스킹 임계치,

는 원신호 마스킹 임계치,

는 분석 계수의 에너지를 나타낸다. here,

Is the error masking threshold,

Is the original signal masking threshold,

Represents the energy of the analysis coefficient.

As shown in Equation 2, the obtained error masking threshold of the linear band corresponds to the partition band again, and the error masking threshold is obtained in the partition band (230). Based on this, if an error masking threshold in a critical band is obtained (240), the error masking threshold in the critical band is used for quantization in the perceptual encoder 140.

The corresponding process from the linear band to the partition band is shown in Equation 3 below.

는 선형 밴드에서의 오차 마스킹 임계치,

는 파티션 밴드에서의 오차 마스킹 임계치를 나타낸다. here,

Is the error masking threshold in the linear band,

Denotes an error masking threshold in the partition band.

)는 동일한 파티션 밴드에 있는 선형 밴드에서의 오차 마스킹 임계치(

) 성분들끼리 합해서 구해진다. Error masking threshold in partition band (

) Is the error masking threshold (in linear bands in the same partition band)

) Are obtained by combining the components.

Next, the error masking threshold? Obtained in the partition band corresponds to the threshold band as in the psychoacoustic model. Like the partition band, the critical band is a nonlinear band that reflects perceptual characteristics, but has fewer total bands than the partition band. Therefore, in order to correspond to the threshold band, the minimum value of the error masking thresholds of the partition bands belonging to the threshold band is selected to be multiplied by the number of bands, and is determined as the error masking threshold of the threshold band. Equation 4 below shows a corresponding relationship from a partition band to a critical band.

는 임계 밴드에서의 오차 마스킹 임계치,

는 파티션 밴드에서의 오차 마스킹 임계치,

와

는 각 임계 밴드에 해당되는 파티션 밴드의 주파수 경계값을 나타낸다.here,

Is the error masking threshold in the critical band,

Is the error masking threshold in the partition band,

Wow

Denotes a frequency boundary value of a partition band corresponding to each threshold band.

3 is a diagram illustrating an embodiment of an input frame in a modified discrete cosine transform process and a psychoacoustic model process applied to the present invention.

In the modified linear prediction encoder 110, the current input frame is input in such a manner that half of the previous input frame and half of the next input frame overlap with each other due to the characteristics of the modified discrete cosine transform (MDCT) (310). This is to enable perfect reconstruction in inverse modified discrete cosine transform (IMDCT). That is, only the section corresponding to 50% of the modified discrete cosine transform is always input to the current input signal (Present Signal) input to the modified linear prediction encoder 110. In addition, only a section corresponding to 50% of the FFT frame for the psychoacoustic model is input.

However, the modified linear prediction encoding unit 110 covers the window (320) so that the training interval of the modified linear prediction encoding further includes a predetermined period before and after the current input frame (320). This is to minimize the interference effect that may appear on the frequency.

As described above, the modified linear prediction encoder 110 obtains an error signal of the current input frame by using the training section (330).

On the other hand, as described above, the psychoacoustic model unit 130 normalizes the original signal masking threshold to the energy of the analysis filter transfer function of the error signal when obtaining the error masking threshold. At this time, a delay occurs between the training interval for obtaining the error signal and the FFT interval for the psychoacoustic model. Therefore, to compensate for this delay, the psychoacoustic model unit 130 normalizes the original signal masking threshold in the linear band by obtaining an average of the transfer function obtained in the previous input frame and the transfer function obtained in the current input frame (340).

)을 취하는 과정이 나타나 있다.Equation 5 below is an average of coefficients of analysis filter coefficients obtained from the previous input frame and analysis filter coefficients obtained from the current input frame.

The process of taking) is shown.

는 이전 입력 프레임에서의 분석필터 계수,

는 현재 입력 프레임에서의 분석필터 계수를 나타낸다.here,

Is the analysis filter coefficients from the previous input frame,

Denotes the analysis filter coefficients in the current input frame.

4 is a configuration diagram of an audio decoding apparatus using modified linear prediction encoding according to the present invention.

As shown in FIG. 4, the audio decoding apparatus according to the present invention includes a bitstream extractor 410, a perceptual decoder 420, an inverse modified discrete cosine transform unit 430, and a modified linear predictive decoder 440. It includes.

The bit stream extractor 410 receives the entire bit stream from the audio encoding apparatus and extracts a bit string corresponding to the current frame from the entire bit stream. The bitstream extractor 410 transfers the extracted bitstream to the decoder 420.

The perceptual decoder 420 analyzes the bit stream extracted by the bit stream extractor 410 to perform an inverse quantization process. That is, the perceptual decoding unit 420 perceptually decodes the bit string to obtain an error signal of the current frame and analysis coefficients of the modified linear prediction coding. The perceptual decoder 420 transfers the obtained error signal of the current frame to the inverse modified discrete cosine transformer 430.

The inverse modified discrete cosine transform unit 430 converts the error signal in the frequency domain to the inverse modified discrete cosine transform to convert the error signal in the time domain. The inverse modified discrete cosine transformer 430 transmits the transformed error signal to the modified linear prediction decoder 440.

The modified linear prediction decoder 440 performs a modified linear prediction decoding process on the error signal in the time domain by using analysis coefficients of the modified linear prediction coding restored by the decoder 420. This decoding process corresponds to an inverse process of the modified linear prediction encoder 110. That is, the modified linear prediction decoder 440 synthesizes an error signal using the analysis coefficients (synthesis coefficients of the synthesis filter) of the modified linear prediction coding to generate a PCM audio signal of a time domain that is a final output.

Hereinafter, an audio encoding method using modified linear prediction coding according to the present invention will be described.

In order to encode a block of a predetermined size in the current frame, the modified linear prediction encoder 110 transforms an error signal by performing a modified linear prediction encoding on a pulse code modulation (PCM) audio signal (original signal) input from the outside in the time domain. Obtain In addition, the modified linear prediction encoder 110 obtains analysis coefficients of the modified linear prediction encoding.

The modified discrete cosine transform unit 120 performs a modified discrete cosine transform on the error signal obtained by the modified linear prediction encoder 110. This is to convert the error signal in the time domain to the frequency domain.

In addition, the first masking threshold calculator 131 calculates a masking threshold for the PCM audio signal (hereinafter, referred to as an original signal masking threshold) using the psychoacoustic model of the PCM audio signal in order to reflect the auditory characteristics of the human frequency domain. Obtain

The second masking threshold calculator 132 masks an error signal in which the perceptual characteristics of the audio signal are reflected using the analysis coefficients of the modified linear prediction coding and the original signal masking threshold values obtained by the first masking threshold calculator 131. The threshold value (hereinafter, referred to as error masking threshold value) is obtained. Next, the second masking threshold calculator 132 calculates a signal-to-masking ratio (hereinafter, error masking SMR) with respect to the error masking threshold using the error masking threshold.

The perceptual encoder 140 performs perceptual encoding on an error signal using an error masking SMR obtained from the psychoacoustic model unit 130 in a modified discrete cosine transform region (frequency domain).

The bitstream packing unit 150 entropy-codes the perceptually encoded error signal by the perceptual encoding unit 140 and packs it into a bit string.

Herein, a process of extracting an error masking threshold value from a masking threshold value of an original signal in an audio encoding method using modified linear prediction coding according to the present invention will be described.

The first masking threshold calculator 131 obtains the original signal masking threshold value from the partition band reflecting the perceptual characteristics without obtaining the masking threshold value from the linear band.

The second masking threshold calculator 132 calculates the original signal masking threshold in the linear band by mapping the original signal masking threshold obtained in the partition band back to the linear band as shown in [Equation 1].

The second masking threshold calculator 132 calculates an error masking threshold in the linear band by normalizing the energy of the analysis coefficients of the modified linear prediction coding using the obtained original signal masking threshold as shown in Equation 2 above.

The second masking threshold calculator 132 calculates the threshold value in the partition band by matching the error masking threshold values in the obtained linear band with the components in the same partition band by adding the components in the same partition band as shown in Equation 3 above. .

The second masking threshold calculation unit 132 selects the minimum value among the error masking thresholds in the partition band as shown in Equation 4, and multiplies the number of bands to obtain the error masking threshold in the threshold band. The second masking threshold calculator 132 calculates an error masking SMR from the obtained error masking threshold in the critical band.

Hereinafter, an audio decoding method using modified linear prediction coding according to the present invention will be described.

The bit stream extractor 410 receives the entire bit stream from the audio encoding apparatus and extracts a bit string corresponding to the current frame from the entire bit stream.

The perceptual decoder 420 analyzes the bit stream extracted by the bit stream extractor 410 to perform an inverse quantization process. That is, the perceptual decoding unit 420 perceptually decodes the bit string to obtain an error signal of the current frame and analysis coefficients of the modified linear prediction coding.

Subsequently, the inverse modified discrete cosine transform unit 430 converts the error signal in the frequency domain to the inverse modified discrete cosine transform to convert the error signal in the time domain.

Thereafter, the modified linear prediction decoder 440 performs a modified linear prediction decoding process on the error signal in the time domain by using analysis coefficients of the modified linear prediction coding restored by the decoder 420. This decoding process corresponds to an inverse process of the modified linear prediction encoder 110. That is, the modified linear prediction decoder 440 synthesizes an error signal using analysis coefficients (synthesis coefficients of the synthesis filter) of the modified linear prediction encoding to generate a PCM audio signal of a time domain that is a final output.

As described above, the method of the present invention may be implemented as a program and stored in a recording medium (CD-ROM, RAM, ROM, floppy disk, hard disk, magneto-optical disk, etc.) in a computer-readable form. Since this process can be easily implemented by those skilled in the art will not be described in more detail.

The present invention described above is capable of various substitutions, modifications, and changes without departing from the technical spirit of the present invention for those skilled in the art to which the present invention pertains. It is not limited by the drawings.

As described above, the present invention can improve the performance of an advanced audio encoder by using an audio encoding structure in which a coding technique through prediction in a time domain and a perceptual encoding in a frequency domain are synthesized, thereby compressing audio encoding. There is an effect that can improve the efficiency.

That is, the present invention compresses a signal by removing the redundancy of an original signal by using modified linear prediction coding, providing an error signal to an audio encoder, and modifying a psychoacoustic model to fit the error signal. There is an effect that the efficiency can be encoded.

In addition, the present invention, by using the modified linear prediction coding having a non-linear frequency resolution in detail to analyze the signal in the low frequency domain in detail to obtain an error signal, the effect of maximizing the coding efficiency through prediction in the time domain There is.

Claims (20)

In an audio encoding apparatus using modified linear prediction encoding, Error signal calculation means for obtaining an error signal by linearly predicting and encoding an audio signal (original signal) input from the outside in a time domain; Frequency domain conversion means for converting the error signal obtained by the error signal calculation means into a frequency domain signal; Masking threshold calculation means for obtaining a masking threshold value used for encoding the error signal using the original signal and encoding information used for linear prediction encoding of the original signal; And Perceptual encoding means for perceptually encoding the error signal converted by the frequency domain transforming means using the obtained masking threshold value. Audio coding apparatus using modified linear prediction coding comprising a. The method of claim 1, Bitstream packing means for coding the perceptually encoded error signal in the perceptual encoding means and packing the result into a bit string Audio encoding apparatus using modified linear prediction coding, further comprising a. The method according to claim 1 or 2, The error signal calculation means, And an error signal obtained by performing Warped Linear Predicted Coding on the original signal in a time domain. The method of claim 3, wherein The masking threshold calculation means, First masking threshold calculation means for obtaining an original signal masking threshold using the psychoacoustic model of the original signal; And Second masking threshold calculation means for obtaining an error masking threshold value by normalizing the obtained original signal masking threshold value by the encoding information, and then obtaining a masking threshold value used for encoding the error signal using the obtained error masking threshold value. Audio coding apparatus using modified linear prediction coding comprising a. The method of claim 4, wherein The second masking threshold calculation means, Linear band threshold calculating means for obtaining an original signal masking threshold in the linear band by matching the original signal masking threshold obtained in the first masking threshold calculating means with a linear band; Linear band error masking threshold calculating means for normalizing the original signal masking threshold in the linear band obtained by the linear band threshold calculating means by the encoding information to obtain an error masking threshold in the linear band; Partition band error masking threshold calculating means for obtaining a threshold in the partition band by matching the error masking thresholds in the linear band obtained by the linear band error masking threshold calculating means with each other by adding the components in the same partition band to the partition band; And The error masking threshold in the critical band is obtained by selecting a minimum value among the error masking thresholds in the partition band obtained by the partition band error masking threshold calculating means and multiplying the number of bands, and encoding the error signal from the error masking threshold in the obtained critical band. Means for calculating a threshold band error masking threshold for calculating a masking threshold used for a Audio coding apparatus using modified linear prediction coding comprising a. The method of claim 5, wherein The masking threshold value used for encoding the error signal is And a signal-to-masking ratio (error masking SMR) with respect to the error masking threshold obtained by the error masking threshold calculating means. The method of claim 6, The perceptual encoding means, And a perceptual encoding of the error signal converted by the frequency domain conversion means by using the error masking SMR. The method of claim 7, wherein The linear band error masking threshold calculation means, An average of the analysis filter coefficients is obtained by averaging the analysis filter coefficients obtained in the previous input frame and the analysis filter coefficients obtained in the current input frame, and using the average of the obtained analysis filter coefficients, the original signal masking threshold is normalized to obtain an error in the linear band. An audio encoding apparatus using modified linear prediction encoding, characterized by obtaining a masking threshold. An audio decoding apparatus using modified linear prediction coding, Bitstream extracting means for extracting a bit string corresponding to a current frame from all bitstreams transmitted from the outside; Perceptual decoding means for perceptually decoding the bit stream extracted by the bit stream extracting means to obtain the error signal of the current frame and the encoding information used for the linear prediction encoding; Error signal conversion means for converting the error signal obtained by the decoding means into an error signal in a time domain; And Modified linear predictive decoding means for generating an audio signal by reconstructing the error signal converted by the error signal converting means using the encoding information used in the linear predictive encoding obtained by the perceptual decoding means Audio decoding apparatus using modified linear prediction coding comprising a. The method of claim 9, The modified linear prediction decoding means, And an error signal converted by the error signal converting means is reconstructed using the analysis coefficients of the modified linear prediction encoding. In the audio coding method using the modified linear prediction coding, An error signal calculating step of obtaining an error signal by linearly predicting and encoding an audio signal (original signal) input from the outside in a time domain; A frequency domain conversion step of converting the error signal obtained in the error signal calculation step into a frequency domain signal; A masking threshold calculation step of calculating a masking threshold value used for encoding the error signal using the original signal and encoding information used for linear prediction encoding of the original signal; And Perceptual encoding step of perceptually encoding the error signal transformed in the frequency domain transforming step using the masking threshold value obtained. An audio encoding method using modified linear prediction coding comprising a. The method of claim 11, A bitstream packing step of coding the perceptually encoded error signal in the perceptual encoding step into a bit string An audio encoding method using modified linear prediction encoding, further comprising. The method according to claim 11 or 12, The error signal calculation step, And an error signal obtained by performing Warped Linear Predicted Coding on the original signal in a time domain. The method of claim 13, The masking threshold calculation step, An original signal masking threshold calculating step of obtaining an original signal masking threshold using the psychoacoustic model of the original signal; And Calculating an error masking threshold value by normalizing the obtained original signal masking threshold value using the encoding information, and then calculating an error masking threshold value used for encoding the error signal using the obtained error masking threshold value. An audio encoding method using modified linear prediction coding comprising a. The method of claim 14, The error masking threshold calculation step, A linear band threshold calculation step of obtaining an original signal masking threshold in the linear band by matching the original signal masking threshold obtained in the original signal masking threshold calculating step with a linear band; A linear band error masking threshold calculating step of obtaining an error masking threshold in a linear band by normalizing the original signal masking threshold in the linear band obtained in the linear band threshold calculating step by the encoding information; A partition band error masking threshold calculating step of calculating a threshold in the partition band by matching the error masking thresholds in the linear band obtained in the linear band error masking threshold calculating step with each component in the same partition band by adding them together; And The error masking threshold in the critical band is obtained by selecting a minimum value among the error masking thresholds in the partition band obtained in the partition band error masking threshold calculation step and multiplying the number of bands, and encoding the error signal from the error masking threshold in the obtained critical band. A threshold band error masking threshold calculation step that yields a masking threshold used for the An audio encoding method using modified linear prediction coding comprising a. The method of claim 15, The masking threshold value used for encoding the error signal is And a signal-to-masking ratio (error masking SMR) with respect to the error masking threshold value obtained in the error masking threshold value calculating step. The method of claim 16, The perceptual encoding step, And a perceptual encoding of the error signal transformed in the frequency domain transforming step using the error masking SMR. The method of claim 17, The linear band error masking threshold calculation step, An average of the analysis filter coefficients is obtained by averaging the analysis filter coefficients obtained in the previous input frame and the analysis filter coefficients obtained in the current input frame. An audio encoding method using modified linear predictive encoding, characterized by obtaining a masking threshold. In the audio decoding method using modified linear prediction coding, A bitstream extraction step of extracting a bit string corresponding to a current frame from all bitstreams transmitted from the outside; A perceptual decoding step of perceptually decoding the bit stream extracted in the bit stream extraction step to obtain the error signal of the current frame and the encoding information used for the linear prediction encoding; An error signal conversion step of converting the error signal obtained in the decoding step into an error signal in a time domain; And A modified linear prediction decoding step of generating an audio signal by reconstructing the error signal converted in the error signal conversion step using the encoding information used in the linear prediction encoding obtained in the perceptual decoding step. Audio decoding method using the modified linear prediction coding comprising a. The method of claim 19, The modified linear prediction decoding step, And reconstructing the error signal converted in the error signal conversion step using the analysis coefficients of the modified linear prediction encoding.

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