JP2007272057A - Audio signal processing apparatus, audio signal processing method, program and recording medium - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To obtain desirable voice for listening, by adding secondary harmonics and interpolating missing harmonic components with voice compression in a simple configuration. <P>SOLUTION: An audio signal processing apparatus of the invention comprises: a first multiplier 112 for multiplying a digital audio signal by a first constant; an adder 114 for adding a second constant to the multiplied digital audio signal; and a second multiplier 116 for multiplying the digital audio signal which is an original signal, by the digital audio signal to which the second constant is added. <P>COPYRIGHT: (C)2008,JPO&INPIT

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 by adding a harmonic having a frequency higher than that of the digital audio signal, a harmonic within the frequency band of the digital audio signal can be obtained. The wave component 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). With this technology, harmonics are generated on a line that is independent of the main signal, and finally added to the main signal. Therefore, the amount of calculation is large, and the cost based on arithmetic processing cannot be reduced. .

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

  The present invention has been made in view of the above-described problems of conventional digital audio signal processing techniques, and an object of the present invention is to add a second harmonic with a simple configuration, and to remove a harmonic component missing due to audio compression. And an audio signal processing apparatus, an audio signal processing method, a program, and a storage medium that can obtain a sound preferable for hearing.

  In order to solve the above problems, according to an aspect of the present invention, a first multiplier that multiplies a digital audio signal by a first constant; an adder that adds a second constant to the digital audio signal multiplied by the constant; There is provided an audio signal processing apparatus comprising: a second multiplier that multiplies the digital audio signal that is an original signal and the digital audio signal to which the second constant is added.

  In the present invention, the addition of the second-order harmonic is realized with a simple configuration without using a Fourier transform or the like, so that low power consumption, high-speed processing, low cost, and reduction in size can be achieved by using a circuit. Can be planned. In addition, since the second harmonic including all components in the frequency band of the digital audio signal is added to the original digital audio signal, the harmonic component missing due to the audio compression is complemented, and the sound preferable for hearing is obtained. Can be obtained.

  A word length extension unit for extending the sample word length of the digital audio signal and transmitting the sample word length to the first multiplier and the second multiplier; and a word length reduction for reducing the sample word length of the signal from the second multiplier And may be further provided.

  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 upsampler that upsamples the digital audio signal and transmits it to the first multiplier and the second multiplier; and a downsampler that downsamples the signal from the second multiplier may be further provided.

  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 up sampler and the down sampler, processing based on a high sampling frequency can be performed in the audio signal processing apparatus.

  A low pass filter (LPF: Low Pass Filter) that limits the high frequency band of the output signal of the second 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 halved of the sampling frequency.

  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 second constant may be represented by a value obtained by subtracting the first constant from 1.

  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.

  A first multiplication step of multiplying the digital audio signal by a first constant using the audio signal processing device; an addition step of adding a second constant to the digital audio signal multiplied by the constant; and the original signal There is also provided an audio signal processing method comprising: a second multiplication step of multiplying the digital audio signal by the digital audio signal added with the second constant.

  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, harmonics in the frequency band of a digital audio signal, in particular, second harmonics can be added to the original digital audio signal with a simple configuration. Therefore, it is possible to complement the overtone component that has been lost due to the audio compression, and it is possible to obtain a sound that is favorable 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 lossy compressed speech such as 3), harmonic components having a low sound pressure are omitted by processing such as Auditory Masking and Perceptual Coding, which are perceptual encoding processing.

  In addition, these compressed speeches 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 audibility evaluation of a conventional vacuum tube amplifier or the like is high, and it is known that a sound containing a large number of even-order harmonics, particularly second-order harmonics, has a favorable effect on audibility. 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 secondary harmonics have received evaluations such as “sound is clean” and “sound is smooth”.

  Based on this background, it has been studied to generate harmonics from the original digital audio signal and to superimpose the harmonics on the digital audio signal to improve the audible sound of the digital audio signal.

  However, in the process aimed at simply extending the frequency band of the digital audio signal, only the harmonics having a frequency higher than the frequency band of the digital audio signal (for example, 16 kHz or higher) among the generated harmonics are added. , 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 technique for adding, this frequency component is deleted 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 improve the audible sound of the violin. .

  There is also 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. However, these technologies are configured in a normal form in which harmonics are generated on a line independent of the main signal and added to the main signal at the end, which inevitably increases the amount of calculation and reduces the cost based on the arithmetic processing. I couldn't.

  In the embodiment of the present invention, with a simple configuration, harmonics missing due to audio compression are added by adding secondary harmonics including almost all components in the frequency band of the digital audio signal to the original digital audio signal. It is possible to complement the components and obtain a sound that is favorable for hearing. Hereinafter, the audio signal processing apparatus of this embodiment will be described in detail.

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全体のゲイン，εは２次高調波のゲインを示し，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。 The configuration for adding the second harmonic to the original digital audio signal as described above 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 of a second harmonic, and G is a numerical value of 1 or less, while ε is 0. It is represented by a small value such as 01.

  FIG. 5 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 three multipliers 12, 14, and 18 and one 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.

(First embodiment: audio signal processing apparatus 100)
However, the audio signal processing apparatus 10 based on Equation 1 requires three multipliers and one adder as described above. The applicant of the present application examined whether it is possible to simply form such an additional configuration of the second harmonic.

数式２を得る。上記ゲインＧは１以下の値なので，Ｇ＝１−δ，Ｇ・ε＝δと置き換えることができる。ただし，

である。上記δは，０．０１〜０．０２といった小さい値を有する。 First, decompose the value in parentheses in Equation 1,

Equation 2 is obtained. Since the gain G is a value of 1 or less, it can be replaced with G = 1−δ and G · ε = δ. However,

It is. The δ has a small value of 0.01 to 0.02.

が生成される。 Thus

Is generated.

  FIG. 1 is a functional block diagram showing a schematic configuration of an audio signal processing apparatus 100 for realizing the derived formula 4. The audio signal processing apparatus 100 includes a word length extension unit 110, a first multiplier 112, an adder 114, a second multiplier 116, and a word length reduction unit 126. The second harmonic generation part of the audio signal processing apparatus 100 according to the present embodiment includes only two multipliers 112 and 116 and one adder 114.

  The audio signal processing apparatus 100 inputs a digital audio signal, performs a process of giving a second harmonic, and outputs the processed signal. 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 the sample word length to the first multiplier 112 and the second multiplier 116. For example, when the input digital audio signal is formed of 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.

で表される。 The first multiplier 112 multiplies the digital audio signal x by a first constant δ to generate δ · x. This constant δ corresponds to the strength of nonlinearity of nonlinear amplification. The relationship between the constant δ and the distortion rate D is

It is represented by

  Since the gain G = 1−δ as described above, the output gain G of the audio signal processing apparatus 100 can be adjusted by changing this δ.

  The adder 114 adds the second constant (1−δ) to the digital audio signal δ · x multiplied by the constant by the first multiplier 112 to generate {(1−δ) + δ · x}. The second constant (1-δ) is inevitably determined when the first constant δ is determined.

  The second multiplier 116 multiplies the digital audio signal, which is the original signal, by the digital audio signal to which the second constant (1-δ) has been added in the adder 114, and x · {(1−δ) + δ. Generate x}. In this way, an output satisfying Equation 4 can be obtained.

FIG. 2 is an explanatory diagram showing frequency-gain characteristics when the audio signal processing apparatus 100 is used. The digital audio signal is limited to a frequency up to ½ of the sampling frequency f _S by the anti-aliasing filter, and its frequency band is f _{L to} f _S / 2 (FIG. 2A). Then, the frequency band is expanded to f _{L to} f _S by the nonlinear filter of y = x {(1−δ) + δx} (FIG. 2B). Here, f _L is an arbitrary low-frequency cutoff frequency.

  The word length reduction unit 126 reduces the sample word length of the signal from the second multiplier 116. 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 126 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 126 can be omitted.

  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. 3 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 first multiplier 112 and the second multiplier 116 (S200).

  The first multiplier 112 multiplies the digital audio signal whose sample word length is extended by a first constant δ (S202), and the adder 114 reads the second constant (1-δ), The digital audio signal multiplied by a constant δ is added (S204).

  Next, the second multiplier 116 multiplies the digital audio signal, which is the original signal, by the digital audio signal to which the second constant is added (S206). Finally, the word length reduction unit 126 performs the second multiplication. The sample word length of the signal from the device 116 is reduced (S208).

  Also by such an audio signal processing method, by adding the second harmonic to the original digital audio signal, it is possible to complement the overtone component missing due to the audio compression, and it is possible to obtain a sound preferable for hearing.

(Second Embodiment: Audio Signal Processing Device 300)
In the second embodiment, the sampling period in the internal processing of the audio signal processing apparatus is increased to increase the calculation processing accuracy such as filter processing.

  FIG. 4 is a functional block diagram showing a schematic configuration of an audio signal processing device 300 according to the second embodiment. The audio signal processing device 300 includes a word length extension unit 110, an upsampler 310, a first multiplier 112, an adder 114, a second multiplier 116, a low-pass filter (LPF) 120, a down-filter. A sampler 314 and a word length reduction unit 126 are included. The second harmonic generation part of the audio signal processing apparatus 300 in this embodiment is also composed of only two multipliers 112 and 116 and one adder 114.

  The word length extension unit 110, the first multiplier 112, the adder 114, the second multiplier 116, and the word length reduction unit 126 already described as the constituent elements in the first embodiment substantially function. Therefore, the up-sampler 310, the low-pass filter 120, and the down-sampler 314 are mainly described here.

  The upsampler 310 upsamples the digital audio signal from the word length extension unit 110 and transmits it to the first multiplier 112 and the second multiplier 116.

This upsampler 310 enables processing based on a high sampling frequency in the audio signal processing apparatus 300. For example, when the up-sampler doubles upsampling, the sampling period f _S is 2 × f _S.

The low pass filter 120 limits the high frequency band of the output signal of the second multiplier 116 and allows only the low frequency band to pass. When nonlinear amplification is performed via the audio signal processing apparatus 100 including the upsampler 310, the frequency component of the original signal is doubled. Therefore, the frequency component of the original signal having the frequency f _S / 4 or more exceeds the Nyquist limit, and the audio signal processing apparatus 100 causes aliasing noise. The low-pass filter 120 finally removes or reduces frequency components exceeding f _S / 2, which become aliasing noise due to the expansion of the frequency band of the upsampler 310, thereby preventing the aliasing noise.

As described above, although the frequency component of f _S / 2 or higher is removed by the low-pass filter 120, the component of the frequency band of the digital audio signal of f _S / 2 or lower remains, and the harmonics of the component itself. Is also generated, so that it is possible to complement the overtone component missing due to the audio compression.

  The downsampler 314 downsamples the signal from the low pass filter 120. For example, when the audio signal processing apparatus 300 functions as a digital audio signal relay apparatus, the input and output sampling periods and sample word lengths must be equal. Here, if the output format of the audio signal processing apparatus 300 can tolerate a high sampling period, the downsampler 314 and the low-pass filter 120 can be omitted as necessary. Further, the down sampler 314 may be integrally formed with the low-pass filter 120 to limit a high-frequency component.

  In the audio signal processing apparatuses 100 and 300 according to the first and second embodiments described above, the second harmonic is added to the digital audio signal, and thereby the harmonic component lost by the audio compression is complemented. The digital audio signal generated by such processing can provide advantageous effects on hearing, such as “clean sound” and “smooth sound”. Further, even if the digital audio signal is not compressed audio, the configuration in which the second harmonic is added can provide a favorable effect on hearing.

  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.

  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.

1 is a functional block diagram showing a schematic configuration of an audio signal processing device according to a first embodiment. It is explanatory drawing which showed the frequency-gain characteristic when using an audio signal processing apparatus. It is the flowchart which showed the flow of the process of the audio signal processing method. It is the functional block diagram which showed the schematic structure of the audio signal processing apparatus by 2nd Embodiment. It is the functional block diagram which showed the schematic structure of the audio signal processing apparatus.

Explanation of symbols

100, 300 Audio signal processing device 110 Word length extension unit 112 First multiplier 114 Adder 116 Second multiplier 120 Low-pass filter (LPF)
126 Word Length Reduction Unit 310 Up Sampler 314 Down Sampler

Claims (9)

A first multiplier for multiplying the digital audio signal by a first constant;
An adder for adding a second constant to the constant-multiplied digital audio signal;
A second multiplier for multiplying the digital audio signal which is an original signal by the digital audio signal to which the second constant is added;
An audio signal processing device comprising: A word length extension unit that extends a sample word length of the digital audio signal and transmits the sample word length to the first multiplier and the second multiplier;
A word length reduction unit for reducing the sample word length of the signal from the second multiplier;
The audio signal processing apparatus according to claim 1, further comprising: An upsampler for upsampling the digital audio signal and transmitting it to the first multiplier and the second multiplier;
A downsampler for downsampling the signal from the second 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 second multiplier.   5. The audio signal processing apparatus according to claim 4, wherein a cutoff frequency of the low-pass filter is ½ of a sampling frequency.   The audio signal processing apparatus according to claim 1, wherein the second constant can be expressed by a value obtained by subtracting the first constant from 1. Computer
A first multiplier for multiplying the digital audio signal by a first constant;
An adder for adding a second constant to the constant-multiplied digital audio signal;
A second multiplier for multiplying the digital audio signal which is an original signal by the digital audio signal to which the second constant is added;
A program characterized by making it function. Computer
A first multiplier for multiplying the digital audio signal by a first constant;
An adder for adding a second constant to the constant-multiplied digital audio signal;
A second multiplier for multiplying the digital audio signal which is an original signal by the digital audio signal to which the second constant is added;
A storage medium that stores a program that allows it to function. A first multiplication step of multiplying the digital audio signal by a first constant;
An adding step of adding a second constant to the digital audio signal multiplied by the constant;
A second multiplication step of multiplying the digital audio signal which is an original signal by the digital audio signal to which the second constant is added;
An audio signal processing method comprising:

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