JP4661667B2 - Audio signal processing apparatus, audio signal processing method, program, and storage medium - Google Patents

Description

  The present invention relates to an audio signal processing apparatus, an audio signal processing method, a program, and a storage medium that process a digital audio signal.

  In recent years, an increase in the processing speed of a computer and an increase in the storage capacity of a memory have made it possible to digitize an analog audio signal once and apply various processes to generate a desired digital audio signal.

  As an example, a technique is known in which harmonics are generated from a digitized digital audio signal and superimposed on the digital audio signal to reproduce a high frequency component of the digital audio signal (for example, Patent Document 1, Patent Document 1). 2).

  However, the above technique is intended to simply extend the frequency band of the digital audio signal, and only by adding a harmonic having a frequency higher than the frequency band of the digital audio signal, The harmonic component in is not added.

  Under such a technique, the overtone component in the frequency band of the digital audio signal omitted in the compressed sound cannot be restored, and depending on the target sound, an effect of being favorable for hearing can be obtained. There wasn't.

  In addition, there is known a technique for adding a harmonic component within the frequency band of a digital audio signal to widen the frequency band and amplify a high-frequency component (for example, Patent Documents 3 and 4). Here again, there remains a problem that the harmonic component in the frequency band of the digital audio signal omitted in the compressed sound cannot be restored.

Japanese Patent Laid-Open No. 2-311006 JP-A-2005-33245 JP-A-5-266582 JP 2005-501278 A

  In order to solve such a problem, an audio signal processing apparatus that generates integer harmonics from a digital audio signal and adds it to the original digital audio signal to restore harmonic components in the frequency band of the digital audio signal has been studied. ing.

  On the other hand, in general, the amplitude of a digital audio signal tends to be large at a low frequency and small at a high frequency. Here, when using the audio signal processing apparatus under investigation to capture a large number of high-order integer high frequencies in the digital audio signal, the high frequency region and the low frequency region are nonlinearly amplified at the same level of amplification. As a result, the amplitude in the low frequency region is clipped.

  The present invention has been made in view of the above-mentioned problems of conventional digital audio signal processing techniques, and an object of the present invention is to add integer order harmonics having different amplification factors depending on the amplitude of the digital audio signal. By providing a new and improved audio signal processing apparatus, audio signal processing method, program, and storage medium capable of complementing the overtone component missing due to the audio compression and obtaining a sound preferable for hearing. is there.

  In order to solve the above-described problem, according to an aspect of the present invention, an adaptive multiplier that nonlinearly amplifies a digital audio signal based on a decreasing function of an even function whose amplification factor decreases as the amplitude increases; A first constant adder for adding a first constant to the output signal of the type multiplier; a signal multiplier for multiplying the output signal of the first constant adder and the digital audio signal; and an output of the signal multiplier There is provided an audio signal processing apparatus comprising: a gain multiplier that adjusts a gain of a signal.

  In the present invention, the addition processing of integer harmonics that guarantees at least the second and higher harmonics can be realized with a simple configuration without using a Fourier transform or the like, so that low power consumption, high speed processing, Cost reduction and downsizing can be achieved by using a circuit. In addition, the harmonic components, mainly the second harmonic, including all components within the frequency band of the digital audio signal, are added to the original digital audio signal. It is possible to obtain a sound that is favorable for hearing.

  In addition, the adaptive multiplier uses a diminishing function of an even function, which is an adaptive nonlinear function, to reduce the nonlinear amplification factor of the harmonics added to the digital audio signal having a large amplitude, and to reduce the digital audio having a small amplitude. It is also possible to increase the nonlinear amplification factor of the harmonics added to the signal. In this case, as a result, the amplification factor in the low frequency region is small, the amplification factor in the high frequency region is large, and it is possible to add a sufficiently large integer harmonic even in the high frequency region. Clipping in the low frequency region can also be avoided.

The adaptive multiplier includes: a square multiplier that squares the digital audio signal; a second constant subtractor that subtracts the output signal of the square multiplier from a second constant; and a second constant subtractor. A third constant multiplier that multiplies the output signal by a third constant; and an output multiplier that multiplies the output signal of the third constant multiplier and the digital audio signal may be provided.

  The adaptive multiplier finally performs even-order nonlinear amplification processing including second-order and fourth-order high frequencies while adjusting the nonlinear amplification factor according to the amplitude of the digital audio signal. By adding a plurality of even-order harmonics, it is possible to further improve audible effects such as “sound is clean” and “sound is smooth”.

The adaptive multiplier includes: an absolute value generator that generates an absolute value of the digital audio signal; a second constant subtractor that subtracts an output signal of the absolute value generator from a second constant; and the second constant subtraction. A third constant multiplier that multiplies the output signal of the unit by a third constant; and an output multiplier that multiplies the output signal of the third constant multiplier and the digital audio signal.

  The adaptive multiplier adjusts the nonlinear amplification factor according to the amplitude of the digital audio signal, and finally performs even-order nonlinear amplification processing represented by second-order, fourth-order,..., Infinite order. The By adding a plurality of even-order harmonics, it is possible to further improve audible effects such as “sound is clean”, “sound is smooth”, and “sound is brilliant”.

  An input stage may further include: a word length extending unit that extends the sample word length of the digital audio signal; and a word length reducing unit that reduces the sample word length of the output signal of the gain multiplier.

  The word length extension unit extends the sample word length (bit length) of the input digital audio signal. Thus, the resolution of the digital audio signal is increased, and the calculation accuracy of the filter processing and the harmonic generation processing can be improved. In addition, since the sample word length is reduced in the word length reduction unit, it can be finally adjusted to the required sample word length of the output.

  An input stage may further include an upsampler that upsamples the digital audio signal; and a downsampler that downsamples the output signal of the gain multiplier.

  The upsampler increases the original sampling frequency to an integral multiple. Conversely, the downsampler lowers the frequency to an integral multiple. With the configuration of the upsampler and downsampler, processing based on a high sampling frequency can be performed in the audio signal processing apparatus.

  A low pass filter (LPF) that limits a high frequency band of the output signal of the gain multiplier may be further provided. Such a low-pass filter is added along with the upsampler. Here, the cutoff frequency of the low-pass filter can be set to ½ of the sampling frequency before upsampling.

  With such a configuration, it is possible to supplement harmonic components in the frequency band of the digital audio signal that is lost due to the audio compression while preventing or reducing aliasing noise generated in the frequency band expanded by upsampling.

  The audio signal processing device is represented by an aggregate of a plurality of components, but each component need not belong to a single device. Further, the above components may function as an electric circuit or a functional module on a computer. Furthermore, the connection between the above components includes a case where they are indirectly connected via other components in addition to the direct connection.

  There are also provided a program that causes a computer to function as the audio signal processing device, and a computer-readable storage medium that stores the program.

  An adaptive multiplication step for nonlinearly amplifying the digital audio signal based on an even function gradually decreasing function whose amplification factor decreases with an increase in amplitude using the audio signal processing apparatus; and an output by the adaptive multiplication step. A first constant adding step for adding a first constant to the signal; a signal multiplying step for multiplying the output signal by the first constant adding step and the digital audio signal; and adjusting the gain of the output signal by the signal multiplying step An audio signal processing method comprising: a gain multiplication step;

  The components corresponding to the dependent claims in the above-described audio signal processing apparatus and the description thereof can be applied to the audio signal processing method, the program, and the storage medium.

  As described above, according to the present invention, integer harmonics that include frequencies within the frequency band of a digital audio signal and that have different amplification factors depending on the amplitude of the digital audio signal are added with a simple configuration. be able to. Therefore, it is possible to complement the harmonic component missing due to the audio compression and obtain a sound preferable for hearing.

  Hereinafter, preferred embodiments of the present invention will be described in detail with reference to the accompanying drawings. In the present specification and drawings, components having substantially the same functional configuration are denoted by the same reference numerals, and redundant description is omitted.

  As shown in "" MPEG digital audio coding "Noll, P .; (Signal Processing Magazine, IEEE, Volume 14, Issue 5, Sept. 1997 Page (s): 59-81)", MP3 (MPEG Audio Layer) In irreversible compressed speech such as 3), harmonic components with low sound pressure are omitted by processes such as Auditory Masking and Perceptual Coding, which are perceptual encoding processes.

  Also, these compressed voices are “Perceptual analysis related to MPEG1-layer3 coding system” Alarcon, SS; (Web Delivering of Music, 2002. WEDELMUSIC 2002. Proceedings. Second International Conference on 9 ~ 11 Dec. 2002, IEEE, Page ( As shown in s): 230) ”, it has received an evaluation that it is inferior in sound quality, especially from a music expert called“ audio file ”, compared to uncompressed speech such as PCM (Pulse Code Modulation). .

  On the other hand, the auditory evaluation of a conventional vacuum tube amplifier or the like is high, and it is known that a sound containing a lot of integer harmonics, particularly second harmonics, has a favorable effect on auditory perception. For example, according to "" The cool sound of tubes [vacuum tube musical applications] "Barbour, E .; (Spectrum, IEEE Volume 35, Issue 8, Aug. 1998 Page (s): 24-35)" Voices that contain complex integer harmonics have received evaluations such as “sound is clean” and “sound is smooth”.

  Under such circumstances, it has been studied to improve the audible sound of the digital audio signal by generating a harmonic from the digital audio signal and superimposing the harmonic on the original digital audio signal.

  However, in the process aimed at simply extending the frequency band of the digital audio signal, only harmonics having a frequency higher than the frequency band of the digital audio signal (for example, 16 kHz or higher) are added among the generated harmonics. Harmonic frequency components within the frequency band of the digital audio signal are not added.

  For example, when the sampling frequency of a digital audio signal is 44.1 kHz, the harmonic (second harmonic) component of the violin sound of the fundamental 2 kHz is 4 kHz, and only harmonics having a frequency higher than the frequency band of the digital audio signal are included. In the above-described conventional technique for adding the frequency component, this frequency component is removed without being added. In compressed audio, harmonic components in the frequency band of such digital audio signals are often omitted, so there is no harmonic component in the end, and it is not possible to obtain an improved audio perception of the violin sound. .

  In the embodiment of the present invention, integer harmonics including almost all components in the frequency band of a digital audio signal, particularly even harmonics mainly including second harmonics are added to the original digital audio signal. Therefore, it is possible to complement the harmonic component missing due to the audio compression and obtain a sound preferable for hearing.

ここで，ｘは入力，ｙは出力，Ｇはオーディオ信号処理装置１０全体の利得，εは２次高調波の利得（増幅率）を示し，Ｇは１以下の数値であるのに対し，εは，０．０１等の小さい値で表される。また，本実施形態を通して，ｘは１以下の実数であり，ｘ＝１となる場合を０ｄＢと定義する。 First, prior to the description of the present embodiment, a configuration for adding a second harmonic to the original digital audio signal will be described. Such digital audio signal processing is realized by Equation 1.

Here, x is an input, y is an output, G is a gain of the entire audio signal processing apparatus 10, ε is a gain (amplification factor) of a second harmonic, and G is a numerical value of 1 or less, while ε Is represented by a small value such as 0.01. Throughout this embodiment, x is a real number equal to or less than 1, and the case where x = 1 is defined as 0 dB.

  FIG. 8 is a functional block diagram showing a schematic configuration of the audio signal processing device 10 for realizing Formula 1. The audio signal processing apparatus 10 includes multipliers 12, 14, and 18 and an adder 16.

First, the multiplier 12 squares the input digital audio signal x, and generates an x ^2. Then, the multiplier 14 multiplies the gain epsilon of the second harmonic generating the epsilon · x ^2.

Then, adder 16, a digital audio signal x is the original signal, adds the second harmonic epsilon · x ^2, generates the ^{(x + ε · x 2)} . Finally, the multiplier 18 can multiply the gain G of the entire digital audio signal to obtain a desired output y.

  When the second harmonic is added to the digital audio signal using such an audio signal processing apparatus 10, the amplification factor of the nonlinear amplifier provided in the audio signal processing apparatus 10 has a uniform value regardless of the amplitude. Take ε. Therefore, the same order of second harmonics are superimposed regardless of the frequency of the digital audio signal.

  On the other hand, in general, the amplitude of a digital audio signal tends to be large at a low frequency and small at a high frequency. Here, using the audio signal processing apparatus 10 to capture a large number of integer-order high frequencies in the high frequency region (region having a small amplitude) of the digital audio signal, as described above, the high frequency region and the low frequency region are the same. As a result, nonlinear amplification is performed at the level amplification factor ε, and as a result, the amplitude in the low frequency region is clipped, and the sound is broken.

  FIG. 9 is an explanatory diagram showing frequency components of a general digital audio signal. Here, the horizontal axis indicates the frequency, the vertical axis indicates the amplitude of the digital audio signal, and the maximum value of the amplitude is 1 (0 dB). Thus, the amplitude of the digital audio signal is often large in the low frequency region and small in the high frequency region.

  FIG. 10 is an explanatory diagram showing frequency components of a digital audio signal to which second harmonics are added using the audio signal processing apparatus 10. As described with reference to FIG. 9, the high frequency region often has a small amplitude, so it is desirable to increase the amplitude of the integer harmonics added to the region. However, when the amplitude of the integer harmonics in such a region is increased, the high frequency region and the low frequency region are nonlinearly amplified with the same level of amplification factor. As a result, the amplitude in the low frequency region is shown in the vicinity of 0 dB in FIG. Clipping will occur.

  In the embodiment of the present invention, in addition to adding an integer order harmonic including almost all components in the frequency band of the digital audio signal to the original digital audio signal, the amplitude of the added integer order harmonic is added. Change the amplification factor according to the amplitude of the digital audio signal, lower the nonlinear amplification factor of the harmonics added to the digital audio signal with a large amplitude, and increase the nonlinear amplification factor of the harmonics added to the digital audio signal with a small amplitude. Yes.

  As described above, since the frequency and amplitude of the digital audio signal are related, changing the amplification factor according to the amplitude of the digital audio signal is equivalent to obtaining a suitable amplification factor for each frequency band. Become. Hereinafter, the audio signal processing apparatus of this embodiment will be described in detail.

(First embodiment: audio signal processing apparatus 100)
Assuming that the nonlinear function of the adaptive multiplier in the first embodiment is ε (x), the response characteristic of the nonlinear amplification portion of the audio signal processing apparatus 100 is expressed by Equation 2.

  FIG. 1 is a functional block diagram showing a schematic configuration of an audio signal processing apparatus 100 for realizing the above mathematical formula 2. The audio signal processing apparatus 100 includes a word length extension unit 110, an upsampler 112, an adaptive multiplier 114, a first constant adder 116, a signal multiplier 120, a gain multiplier 122, a low-pass signal. A filter 124, a down sampler 126, and a word length reduction unit 128 are included.

  The audio signal processing apparatus 100 receives a digital audio signal, performs processing for adding an integer order harmonic to the original signal, and outputs the result. Such a digital audio signal is an audio signal output from a digital audio player such as a CD, MD, MP3 player, or an arbitrary digital audio signal processing circuit. The digital audio signal is not limited to the above signal, and may be an analog audio output from an analog record player or a cassette deck, or an audio obtained by A / D converting an analog signal from an analog audio output of a digital audio player.

  The word length extension unit 110 extends the sample word length of the digital audio signal input from the outside of the audio signal processing apparatus 100 and transmits it to the upsampler 112. When the input digital audio signal is formed with 16 bits, the word length is extended to 32 bits, for example. Thus, the resolution of the digital audio signal is increased, and the calculation accuracy in the subsequent multiplier can be improved.

  The upsampler 112 upsamples the digital audio signal from the word length extension unit 110 and transmits it to the adaptive multiplier 114 and the signal multiplier 120, respectively. As the upsampler 112, an interpolation filter such as a polyphase filter can be used.

The upsampler 112 enables processing based on a sampling frequency higher than that of the original signal in the audio signal processing apparatus 100. For example, if the upsample rate of the upsampler 112 is doubled, the sampling frequency f _S of the original digital audio signal is upsampled to 2 × f _S. If the sampling frequency has already been expanded in the previous stage of the upsampler 112, the upsample 112 can be omitted. Here, the sampling frequency of the digital audio signal refers to the sampling frequency when the audio signal is converted from an analog signal to a digital signal.

  The adaptive multiplier 114 nonlinearly amplifies the digital audio signal based on an adaptive nonlinear function ε (x). Here, “adaptive” means changing the count multiplied by x according to the amplitude value of the input digital audio signal x. Therefore, the nonlinear function ε (x) is expressed by a function of x.

  The adaptive multiplier 114 can change the amplitude of the integer harmonics to be added according to the amplitude of the digital audio signal, and the integer harmonics in an arbitrary frequency region where the digital audio signal has a specific amplitude can be obtained. It becomes possible to emphasize.

The adaptive multiplier 114 may be represented by an even function gradual decreasing function in which the amplification factor decreases as x increases. In such a gradual decrease function, the nonlinear amplification factor of the harmonic added to the digital audio signal having a large amplitude is lowered, and the nonlinear amplification factor of the harmonic added to the digital audio signal having a small amplitude is raised. Specifically, as x increases from 0 to 1, the amplification factor gradually decreases from an arbitrary constant ε ₀ to 0. The arbitrary constant ε ₀ may be set individually by the user or may be preset.

  Using such a decreasing function of the even function results in a small non-linear amplification factor of integer harmonics in the low frequency region and a large non-linear amplification factor of integer harmonics in the high frequency region. In addition, it is possible to add a sufficiently large integer harmonic, and also avoid clipping in the low frequency region.

  The first constant adder 116 adds “1” which is the first constant 118 to the output signal ε (x) · x of the adaptive multiplier 114.

  The signal multiplier 120 multiplies the output signal (1 + ε (x) · x) at the final stage of the adder 116 and the digital audio signal x that is the original signal to derive x (1 + ε (x) · x). To do.

  The gain multiplier 122 multiplies the output signal x (1 + ε (x) · x) of the signal multiplier 120 by the gain G adjusted so that the output of the gain multiplier 122 does not cause gain over. In this way, an output satisfying Equation 2 can be obtained.

The low-pass filter 124 limits the high frequency band of the output signal of the gain multiplier 122 and passes only the low frequency band. When nonlinear amplification is performed via the audio signal processing apparatus 100 including the upsampler 112, the frequency component of the original signal becomes high. For example, when the harmonic is a second-order harmonic, the frequency component of the original signal having a frequency f _S / 4 or higher is the same, and when the harmonic is a third-order harmonic, the frequency component of the frequency f _S / 6 or higher is similarly n-order In the case of a higher harmonic, a frequency component of f _S / 2n or more exceeds the Nyquist limit, resulting in aliasing noise.

  In order to prevent this aliasing noise, it is necessary to perform upsampling of N times or more, where N is the highest order of nonlinear amplification. Therefore, the upsampler 112 must satisfy the upsampling rate L such that L ≧ N (L and N are integers).

Further, the low-pass filter 124 finally removes or reduces the frequency component exceeding f _S / 2, which becomes aliasing noise, in accordance with the upsampling in the frequency band of the upsampler 112, and reduces aliasing noise. Suppress. The cut-off frequency f _S / 2 is a case where the sampling frequency before up-sampling is f _S. Depending on the sampling frequency up-sampled at this time, the low-pass filter 124 can be omitted.

When the low-pass filter 124 is implemented, the frequency component of f _S / 2 or higher is removed by the low-pass filter 124, but the frequency band component of the digital audio signal of f _S / 2 or lower remains, and Since harmonics of the component itself are also generated, it is possible to supplement the overtone component missing due to the audio compression.

  The downsampler 126 downsamples the signal from the low-pass filter 124. For example, when the audio signal processing device 100 functions as a digital audio signal relay device, the input and output sampling frequencies must be equal. Here, if the output format of the audio signal processing apparatus 100 can tolerate a high sampling frequency, the downsampler 126 can be omitted as necessary. Further, the down sampler 126 may be formed integrally with the low-pass filter 124 to limit the high-frequency component.

  The word length reduction unit 128 reduces the sample word length of the signal from the downsampler 126. As described above, the word length extension unit 110 extends the sample word length of the digital audio signal input from the outside of the audio signal processing apparatus 100. Therefore, the sample word length processed in the audio signal processing apparatus 100 may differ from the required output sample word length. The word length reduction unit 128 reduces the sample word length in accordance with the output format. In addition, when the output destination allows the extended sample word length, the word length reduction unit 128 can be omitted.

  The audio signal processing apparatus 100 according to the present embodiment can change the amplitude of the integer harmonics to be added according to the amplitude of the digital audio signal by nonlinear amplification by the adaptive multiplier 114. Therefore, by adding an integer harmonic that includes a frequency within the frequency band of the digital audio signal and has an amplification factor that differs depending on the amplitude of the digital audio signal, a harmonic that has been lost due to audio compression is obtained. Complementation of ingredients is possible.

  Specifically, the purpose is to process the voice by newly creating and adding a harmonic component that has been lost due to the voice compression, and in particular to bring it closer to the original voice before the voice compression. In general, it is said that voices with even-order harmonics are warm or soft and comfortable, and voices with odd-order harmonics are bright and sharp. Therefore, by properly balancing the harmonics of the integer order, for example, it is possible to generate a sound peculiar to a vacuum tube by a vacuum amplifier.

  Each component described above may be formed by an electric circuit (hardware) or may function as a functional module (software) on a computer. Also provided are a program that causes a computer to function as the audio signal processing apparatus 100 and a computer-readable storage medium that stores the program.

(Audio signal processing method)
An audio signal processing method for processing a digital audio signal using the above-described audio signal processing apparatus 100 is also provided.

  FIG. 2 is a flowchart showing a processing flow of the audio signal processing method. First, the audio signal processing apparatus 100 extends the sample word length of the input digital audio signal and transmits it to the upsampler 112 (S200).

  Subsequently, the up-sampler 112 up-samples the digital audio signal whose sample word length is extended and transmits the digital audio signal to the adaptive multiplier 114 and the signal multiplier 120, respectively (S202). The adaptive multiplier 114 nonlinearly amplifies the upsampled digital audio signal based on the nonlinear function ε (x) (S204).

  The first constant adder 116 adds “1”, which is the first constant 118, to the output signal ε (x) · x of the adaptive multiplier 114 (S206), and the signal multiplier 120 The output signal 116 (1 + ε (x) · x) is multiplied by the digital audio signal x which is the original signal (S208).

  Next, the gain multiplier 122 multiplies the output signal of the signal multiplier 120 by the gain G adjusted so that the output of the gain multiplier 122 does not cause gain over (S210), and the low-pass filter 124 The high frequency band of the output signal of the gain multiplier 122 is limited (S212).

  The down sampler 126 down-samples the signal from the low-pass filter 124 (S214). Finally, the word length reduction unit 128 uses the sample word length extended by the word length extension unit 110 as the original word length. (S216).

  Even with such an audio signal processing method, integer harmonics having different amplification factors depending on the amplitude of the digital audio signal are added to the original digital audio signal, thereby enhancing a specific frequency region and missing by audio compression. It is possible to complement the overtone component thus obtained, and it is possible to obtain a sound that is favorable for hearing.

(Second Embodiment: Adaptive Multiplier 300)
In the second embodiment, another embodiment of the adaptive multiplier 114 will be described in detail.

  FIG. 3 is a functional block diagram showing a schematic configuration of the adaptive multiplier 300 according to the second embodiment. The adaptive multiplier 300 includes a square multiplier 310, a second constant subtractor 312, a third constant multiplier 316, and an output multiplier 318.

  The square multiplier 310 multiplies the digital audio signal as the original signal to generate a second harmonic.

The second constant subtractor 312 subtracts the output signal x ² of the square multiplier 310 from “1” which is the second constant 314.

The third constant multiplier 316 multiplies the output signal (1-x ² ) of the second constant subtractor 312 by the third constant ε ₀ .

The output multiplier 318 multiplies the output signal ε ₀ (1-x ² ) of the third constant multiplier 316 and the digital audio signal x.

となる。 The output signal of the adaptive multiplier 300 is

It becomes.

FIG. 4 is an explanatory diagram showing nonlinear amplification by the adaptive multiplier 300. Referring to FIG. 4, in the nonlinear amplification curve represented by Equation 3, the amplification factor gradually decreases from an arbitrary constant ε ₀ to 0 as x increases from 0 to 1 or x decreases from 0 to −1. It is understood that it is a decreasing function. Accordingly, it is possible to reduce the non-linear amplification factor of the integer harmonic of the digital audio signal having a large amplitude and increase the non-linear amplification factor of the integer harmonic of the digital audio signal having a small amplitude.

となる。これは，適応型乗算器３００を用いた場合のオーディオ信号処理装置１００の応答を示す。数式４を参照すると，適応型乗算器３００を用いたオーディオ信号処理装置１００は，２次の高調波と４次の高調波とによる偶数次の高調波が付加されることが分かる。 Substituting Equation 3 above into Equation 2,

It becomes. This shows the response of the audio signal processing apparatus 100 when the adaptive multiplier 300 is used. Referring to Equation 4, it can be seen that the audio signal processing apparatus 100 using the adaptive multiplier 300 adds even-order harmonics due to the second-order harmonics and the fourth-order harmonics.

  By adding such higher-order harmonics, it is possible to further improve the audible effects such as “sound is clean” and “sound is smooth”.

(Third Embodiment: Adaptive Multiplier 400)
In the third embodiment, another embodiment of the adaptive multiplier 114 will be described in detail.

  FIG. 5 is a functional block diagram showing a schematic configuration of the adaptive multiplier 400 according to the third embodiment. The adaptive multiplier 400 includes an absolute value generator 410, a second constant subtractor 412, a third constant multiplier 416, and an output multiplier 418.

  The absolute value generator 410 generates an absolute value | x | of the digital audio signal x as an original signal.

  The second constant subtractor 412 subtracts the output signal | x | of the absolute value generator 410 from “1” which is the second constant 414.

The third constant multiplier 416 multiplies the output signal (1− | x |) of the second constant subtractor 412 by the third constant ε ₀ .

The output multiplier 418 multiplies the output signal ε ₀ (1− | x |) of the third constant multiplier 416 and the digital audio signal x.

となる。 The output signal of the adaptive multiplier 400 is

It becomes.

FIG. 6 is an explanatory diagram showing nonlinear amplification by the adaptive multiplier 400. Referring to FIG. 6, the nonlinear amplification curve expressed by Equation 5 shows a gradual decrease in the amplification factor that gradually decreases from an arbitrary constant ε ₀ to 0 as x increases from 0 to 1 or decreases from 0 to −1. It is understood that it is a function. Accordingly, it is possible to reduce the non-linear amplification factor of the integer harmonic of the digital audio signal having a large amplitude and increase the non-linear amplification factor of the integer harmonic of the digital audio signal having a small amplitude.

となる。これは，適応型乗算器４００を用いた場合のオーディオ信号処理装置１００の応答を示す。 Substituting Equation 5 above into Equation 2,

It becomes. This shows the response of the audio signal processing apparatus 100 when the adaptive multiplier 400 is used.

と表される。ここでｂ_ｋはフーリエ級数展開で決定される定数である。 When Equation 6 is specifically examined, | x | can be considered as a triangular wave having an interval [−1, 1] of x as one cycle. Here, when Fourier series expansion is used for | x |, since | x | is an even function,

It is expressed. Here, b _k is a constant determined by Fourier series expansion.

を適用すると，

となる。 In Equation 7,

Applying

It becomes.

で表される特定の定数である。 Where _cn is

It is a specific constant represented by

となる。この数式１１を参照すると，適応型乗算器３００を用いたオーディオ信号処理装置１００は，無限の偶数次の高調波が付加されることが分かる。 Furthermore, substituting the above equation 9 into equation 6,

It becomes. Referring to Equation 11, it can be seen that the audio signal processing apparatus 100 using the adaptive multiplier 300 adds infinite even-order harmonics.

FIG. 7 is an explanatory diagram showing the frequency-amplitude characteristics of the audio signal processing apparatus 100 expressed by Equation 11. Here, the horizontal axis indicates the frequency, the vertical axis indicates the amplitude of the digital audio signal, and the amplitude of the fundamental frequency is 1 (0 dB). In such Fig. 7, the audio signal processing apparatus 100 is understood that infinite even-order harmonics can be obtained, The amplitude than the secondary harmonic fourth order harmonic increases by an arbitrary constant c _n It is also understood.

  By adding such even-order harmonics, it is possible to further improve audible effects such as “sound is clean”, “sound is smooth”, and “sound is brilliant”.

  In the audio signal processing apparatus 100 according to the second and third embodiments described above, even-order harmonics are added to the digital audio signal, thereby complementing the harmonic component lost by the audio compression. The digital audio signal generated by such processing can provide a favorable effect on hearing. Further, even if the digital audio signal is not compressed audio, an advantageous effect in terms of audibility can be obtained by adding the integer harmonics.

  In addition, since the audio signal processing apparatus according to the above-described embodiment can basically be configured by only a digital filter and a memory, the amount of processing is small compared to a case where a complicated algorithm using Fourier transform is applied. The objectives can be achieved by the system, which is very advantageous in terms of required system resources and implementation costs.

  As mentioned above, although preferred embodiment of this invention was described referring an accompanying drawing, it cannot be overemphasized that this invention is not limited to the example which concerns. It will be apparent to those skilled in the art that various changes and modifications can be made within the scope of the claims, and these are naturally within the technical scope of the present invention. Understood.

  For example, in the above-described embodiment, a computer capable of performing filter processing at high speed is taken as an example, but the present embodiment is also applied to relatively small and inexpensive digital audio equipment such as a portable audio player and system stereo. It is possible. In this case, even without using an expensive and large-capacity vacuum tube amplifier or the like, it is possible to obtain a timbre with favorable characteristics similar to a vacuum tube amplifier such as “clean” and “smooth” only by adding a relatively small digital circuit. . In addition, since it can be applied to all devices using digital audio in principle, it can be applied to an extremely wide range of electronic devices such as car audio, digital television, and home theater system.

In the above-described embodiment, (1-x ² ) and (1- | x |) are described as the nonlinear functions of the adaptive multiplier, but as x increases from 0 to 1, If the amplification factor is an even function gradually decreasing function that gradually decreases from an arbitrary constant ε ₀ to 0, the present invention is not limited to such a case. For example, ε ₀ cos (x), ε ₀ (1-x ⁴ ), ε ₀ (1 Various nonlinear functions such as −x ⁶ ) can be applied.

  Note that the steps in the audio signal processing method of the present specification do not necessarily have to be processed in chronological order according to the order described in the flowchart, but are performed in parallel or individually (for example, parallel processing or object processing). May be included.

It is the functional block diagram which showed the schematic structure of the audio signal processing apparatus for implement | achieving Numerical formula 2. FIG. It is the flowchart which showed the flow of the process of the audio signal processing method. FIG. 6 is a functional block diagram illustrating a schematic configuration of an adaptive multiplier according to a second embodiment. It is explanatory drawing which showed the nonlinear amplification by an adaptive multiplier. FIG. 9 is a functional block diagram illustrating a schematic configuration of an adaptive multiplier according to a third embodiment. It is explanatory drawing which showed the nonlinear amplification by an adaptive multiplier. It is explanatory drawing which showed the frequency amplitude characteristic of the audio signal processing apparatus represented by Numerical formula 11. 2 is a functional block diagram illustrating a schematic configuration of an audio signal processing device for realizing Formula 1. FIG. It is explanatory drawing which showed the frequency component of the general digital audio signal. It is explanatory drawing which showed the frequency component of the digital audio signal which added the 2nd harmonic using the audio signal processing apparatus.

Explanation of symbols

10, 100 Audio signal processor 110 Word length extension unit 112 Upsampler 114, 300, 400 Adaptive multiplier 116 First constant adder 120 Signal multiplier 122 Gain multiplier 124 Low pass filter 126 Downsampler 128 Word length reduction Unit 310 square multipliers 312, 412 second constant subtractors 316, 416 third constant multipliers 318, 418 output multiplier 410 absolute value generator

Claims (10)

An adaptive multiplier that non-linearly amplifies the digital audio signal based on an evenly decreasing function whose gain decreases with increasing amplitude;
A first constant adder for adding a first constant to the output signal of the adaptive multiplier;
A signal multiplier for multiplying the output signal of the first constant adder and the digital audio signal;
A gain multiplier for adjusting the gain of the output signal of the signal multiplier;
An audio signal processing device comprising: The adaptive multiplier is
A square multiplier for squaring the digital audio signal;
A second constant subtractor for subtracting the output signal of the square multiplier from a second constant;
A third constant multiplier for multiplying an output signal of the second constant subtractor by a third constant;
An output multiplier for multiplying the output signal of the third constant multiplier and the digital audio signal;
The audio signal processing apparatus according to claim 1, further comprising: The adaptive multiplier is
An absolute value generator for generating an absolute value of the digital audio signal;
A second constant subtractor for subtracting the output signal of the absolute value generator from a second constant;
A third constant multiplier for multiplying an output signal of the second constant subtractor by a third constant;
An output multiplier for multiplying the output signal of the third constant multiplier and the digital audio signal;
The audio signal processing apparatus according to claim 1, further comprising: A word length extension for extending the sample word length of the digital audio signal at the input stage;
A word length reduction unit for reducing the sample word length of the output signal of the gain multiplier;
The audio signal processing apparatus according to claim 1, further comprising: An upsampler for upsampling the digital audio signal at an input stage;
A downsampler for downsampling the output signal of the gain multiplier;
The audio signal processing apparatus according to claim 1, further comprising:   The audio signal processing apparatus according to claim 1, further comprising a low-pass filter that limits a high frequency band of an output signal of the gain multiplier.   The audio signal processing apparatus according to claim 6, wherein a cutoff frequency of the low-pass filter is ½ of a sampling frequency of the digital audio signal. Computer
An adaptive multiplier that non-linearly amplifies the digital audio signal based on an evenly decreasing function whose gain decreases with increasing amplitude;
A first constant adder for adding a first constant to the output signal of the adaptive multiplier;
A signal multiplier for multiplying the output signal of the first constant adder and the digital audio signal;
A gain multiplier for adjusting the gain of the output signal of the signal multiplier;
A program characterized by making it function. Computer
An adaptive multiplier that non-linearly amplifies the digital audio signal based on an evenly decreasing function whose gain decreases with increasing amplitude;
A first constant adder for adding a first constant to the output signal of the adaptive multiplier;
A signal multiplier for multiplying the output signal of the first constant adder and the digital audio signal;
A gain multiplier for adjusting the gain of the output signal of the signal multiplier;
A storage medium that stores a program that allows it to function. An adaptive multiplication step for non-linearly amplifying the digital audio signal based on an evenly decreasing function whose gain decreases with increasing amplitude;
A first constant addition step of adding a first constant to the output signal by the adaptive multiplication step;
A signal multiplication step of multiplying the output signal from the first constant addition step by the digital audio signal;
A gain multiplication step of adjusting the gain of the output signal by the signal multiplication step;
An audio signal processing method comprising:

Images