JPH1168494A - Audio signal processor and storing medium recording audio signal processing program - Google Patents

Abstract

PROBLEM TO BE SOLVED: To naturalize the connection of sounds and prevent head omission by obtaining a level control gain derived from approximation through a prescribed function which is saturated with a level value control gain in an area not smaller than an upper limit value in a prescribed threshold value range, and multiplying an input signal the obtd. level control gain. SOLUTION: In the area where the level of an input digital audio signal is a threshold Vth1 to Vth2 , a level control gain approximate-processed by a m-th order function (m>2) saturated with an area control gain Gm not smaller than a threshold value Vth1 is obtained. Namely, a device multiplying the gain Gm by a digital audio signal makes Gm=1 in an area where the level of an input signal is not lower than a threshold value Vth1 and makes Gm=(Vth2 / Vth1 )<n-1> in an area where the level of the input signal is not higher than a threshold value Vth2 . Then an expanding device multiplying a level control gain Gm approximate-processed by the m-th order function saturated with the gain Gm by the digital audio signal in an area where the level of the input signal is a threshold value Vth1 -Vth2 is provided. The change of an output level near Vth1 is smoothed to prevent the head omission of sounds near Vth2 .

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an education / athletic facility,
The present invention relates to an audio signal processing device and a storage medium storing an audio signal processing program for automatically controlling a speaker's loud voice level in a multipurpose hall, a banquet hall, and the like to an optimum level, and particularly to an expanding process and a compressing process.

[0002]

2. Description of the Related Art Generally, as a method of automatically controlling a speaker's loud voice level to an optimum level, a gain is increased from "1" (expansion: expanding) when an off microphone (low voice) is used, and an excessively high on microphone (loud voice) is used. In some cases, the gain is reduced from "1" (compression: compression), and when an impact sound such as a microphone drop occurs, the gain is further reduced (suppression: limit).

[0003] As a conventional expanding method, a downward type and an upward type are known. FIG. 12 shows the conventional downward expansion characteristics and compression characteristics. The horizontal axis indicates the input audio level Xi [dB], and the vertical axis indicates the output audio level, Yo [dB]. However, Xi [dB] = 20 log ₁₀ xi [V] Yo [dB] = 20 log ₁₀ yo [V] Two threshold values Vth1 and Vth2 in the expanded area.
[DB] (Vth1> Vth2), and the elongation rate = n
(1 ≦ n ≦ ∞), the logarithmic input / output characteristic is Xi ≧ V
In the region of th1, Yo = Xi (1) In the region of Vth2 <Xi <Vth1, Yo = Vth1-n (Vth1-Xi) (2) In the region of Xi.ltoreq.Vth2, Yo = Xi- (n-1) (Vth1 −Vth2) (3), all of which are linear expressions.

Therefore, if the threshold values vth1 [V] and vth2 [V] of xi [V], yo [V] and xi [V] are set to Li = | xi | Lo = | yo | vth1> 0 vth2> 0, then Li Lo = Li in the region of ≧ vth1 (4) Lo = Li · (Li / vth1) ^{n−1 in} the region of vth2 <Li <vth1 (5) Lo = Li · (vth2 / vth1 in the region of Li ≦ vth2 ) ^N-1 (6)

If the gain between input and output is Gm, Li
Gm = 1 in the region of .gtoreq.vth1 (7) Gm = (Li / vth1) ^{n-1 in} the region of vth2 <Li <vth1 Gm = (vth2 / vth1) ^{n-1 in} the region of Li.ltoreq.vth2 ... (9)

Here, in order to digitally process the gain Gm having such characteristics, a digital signal processor (DSP) is usually used to obtain the equations (7), (8), and (9) according to the input signal area. It is necessary to perform arithmetic processing based on In this case, equation (9) can be uniquely calculated if vth1 and vth2 are given, and it is not necessary to perform power calculation by the DSP. However, since equation (8) needs to perform exponentiation calculation, straight-line approximation processing is generally used in the conventional method. For example, the paper “S. Wei etc, J. Acoust. Soc. Jpn. (E) 16,
6 (1995) -Realization of dinamic range controllers
on a digital signal processor for audio systems ''
Is known as a linear approximation of a polynomial using an m-th power series.

FIG. 13 shows an upward expansion characteristic as another conventional example. Here, as shown in FIG. 12, in the case of the downward type, the gain becomes 1 or less in an area where the gain is equal to or less than the threshold value Vth1, and in the case of the upward type, the gain becomes the threshold value Vth1.
It becomes 1 or more in the region of th2 or more. Therefore, this gain characteristic Gm is Gm = (vth1 / vth2) ^{n-1 in} the region of Li≥vth1 (10) Gm = (Li / vth2) ^{n-1 in} the region of vth2 <Li <vth1 (11) Li Gm = 1... (12) in the region of .ltoreq.vth2. Therefore, in the case of the upward type, similarly to the downward type, since the gain characteristic Gm is a linear expression, when processing is performed by the DSP, the equation (11) is subjected to linear approximation processing.

FIG. 12 further shows a conventional compression process in which the gain is reduced from "1" when an excessively large on-mic (loud voice) occurs. This processing is usually performed after the above-described expanding processing, and is performed with a saturation characteristic or a linear characteristic in which the gain gradually decreases from "1" in a region equal to or larger than the threshold value Vth-comp (> Vth1).

[0009]

However, FIG.
2. The conventional expand characteristic shown in FIG. 13 is a linear expression, and the output level does not change smoothly with respect to the input level near the threshold values Vth1 and Vth2. The problem becomes more pronounced in the vicinity because the volume is high. Further, referring to the level change on the linear axis in the linear characteristic on the logarithmic axis, the rising change of the sound at the time of level extension is gentle near the low threshold value Vth2 and rapidly changes near the high threshold value Vth1. Therefore, the time until the expander is activated after a signal larger than the threshold is input (attack time)
Depending on the length, there is a problem that a so-called "sound truncation" occurs near the low threshold value Vth2. Furthermore, when performing linear approximation processing for exponentiation calculation using a DSP, there is a problem that the number of signal processing steps of the DSP is increased if an attempt is made to increase the calculation accuracy.

In addition, the saturation type compression characteristic as shown in FIG. 12 has a problem that inflection disappears in a loud voice region and the sound becomes muffled. In the loud voice region, it is desirable that the sound has inflection and is sharp and easy to hear.

In view of the above-mentioned conventional problems, the present invention makes it possible to smoothly change the output level with respect to the input level near the high threshold value Vth1 in the case of performing the expanding process, to make the sound connection natural, and to reduce the low threshold value. Vth2
Provided is an audio signal processing device capable of improving a fall characteristic of a sound in the vicinity thereof, preventing a sound from being cut off, and further reducing the number of signal processing steps when performing processing using a DSP. With the goal.

[0012]

According to the present invention, in order to achieve the above object, in the region where the level of the input digital audio signal is in the range of the threshold value Vth1−Vth2, the level control gain Gm in the region in which the level is equal to or larger than the threshold value Vth1 is saturated with the mth order. The level control gain Gm is approximated by a function (m is an integer of 2 or more). That is, according to the present invention, in an audio signal processing device that expands by multiplying a digital audio signal by a level control gain Gm, Gm = 1 is set in an area where the level of the input digital audio signal is equal to or higher than the threshold value Vth1, and Audio signal level is threshold Vth
Gm = (Vth2 / Vth1) ^{n-1 in} the area of 2 or less, and the level of the input digital audio signal becomes the threshold Vth
Expanding means for multiplying the digital audio signal by a level control gain Gm approximated by an m-th order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the area above the threshold Vth1 in the area of 1-Vth2,
An audio signal processing device is provided.

Further, according to the present invention, in an audio signal processing apparatus for expanding by multiplying a digital audio signal by a level control gain Gm, thresholds for determining the level of an input digital audio signal are set to Vth1, Vth2 (Vth1). > Vth2), the expand ratio is n, and in the region where the level of the input digital audio signal is equal to or higher than the threshold value Vth1, Gm = (Vth1 / Vth2) ^n−1, and the level of the input digital audio signal is the threshold value Vth.
Gm = 1 in the region below 2 and the level of the input digital audio signal is
Expanding means for multiplying the digital audio signal by a level control gain Gm approximated by an m-th order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the area above the threshold Vth1 in the area of 1-Vth2,
An audio signal processing device is provided.

Further, according to the present invention, the threshold for determining the level of the input digital audio signal is set to V for the audio signal processing to be expanded by multiplying the digital audio signal by the level control gain Gm.
th1 and Vth2 (Vth1> Vth2), the expand ratio is n, and the level of the input digital audio signal is Gm = 1 in the region where the level of the input digital audio signal is equal to or higher than the threshold Vth1, and the level of the input digital audio signal is the threshold Vth.
Gm = (Vth2 / Vth1) ^{n-1 in} the area of 2 or less, and the level of the input digital audio signal becomes the threshold Vth
In the region of 1-Vth2, there is an expanding step of multiplying the digital audio signal by a level control gain Gm approximated by an m-order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the region above the threshold Vth1. A storage medium on which an audio signal processing program is recorded is provided.

Further, according to the present invention, the threshold for determining the level of the input digital audio signal is set to V for the audio signal processing to be expanded by multiplying the digital audio signal by the level control gain Gm.
th1 and Vth2 (Vth1> Vth2), the expand ratio is n, and in the region where the level of the input digital audio signal is equal to or higher than the threshold value Vth1, Gm = (Vth1 / Vth2) ^n-1 and the level of the input digital audio signal is the threshold value Vth.
Gm = 1 in the region below 2 and the level of the input digital audio signal is
In the region of 1-Vth2, there is an expanding step of multiplying the digital audio signal by a level control gain Gm approximated by an m-order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the region above the threshold Vth1. A storage medium on which an audio signal processing program is recorded is provided.

[0016]

Embodiments of the present invention will be described below with reference to the drawings. FIG. 1 is an explanatory diagram showing an algorithm of an expander which is a main part of an audio signal processing apparatus according to an embodiment of the present invention, FIG. 2 is a graph showing characteristics of the expander of FIG. 1, and FIG. It is a flowchart for demonstrating the process of a panda.

The expander shown in FIG. 1 is configured to multiply an input level Xi by a first expanded gain Gm (MPY shown) and then multiply the multiplied value by a second expanded gain Ge. The first expanded gain Gm is, as shown in FIG.
In the region of ^{Gm = (Vth2 / Vth1) n} -1 Vth2 <Xi <Vth1 is th2 region, is controlled to be Gm = 1 in the region of the Gm = 1-k (Δ + qΔ 2) / Li Xi ≧ Vth1 . Therefore, according to such characteristics, the sound is connected so that the output level near the high threshold value Vth1 becomes smooth with respect to the input level, and the sound fall characteristic near the low threshold value Vth2 is improved. Can be prevented from breaking.
Here, in the region of Xi ≦ Vth2, Gm = constant value <1, and the region is of a downward expansion type. Therefore, in the region, the region is of the acceleration saturation characteristic of the downward type.

Here, the m-order function of the area is quadratic, the difference Δ between the threshold value Vth1 and the absolute value of the input level Xi is Δ = vth1− | xi | = vth1−li, and the characteristic conversion is performed. The function f (Δ) having Δ as a variable is expressed as f = k (Δ + qΔ ² ) (13) where k and q are approximated by quadratic expressions like parameters, and outputs Yo and Lo are input. Xi and the function f are given by the following equations. yo = xi−f (xi> 0) (14) yo = xi + f (xi ≦ 0) (15)

[0019] Expressing this in absolute expression, Lo = Li {1-k (Δ + qΔ 2) / Li} ... (16) becomes, therefore, the gain Gm between input and output Gm = 1-k (Δ + qΔ 2) / Li (17)

Next, the expander processing will be described in detail with reference to FIGS. First, when an audio signal xi having positive and negative polarities is input (step S1), Li = | xi | is obtained by rectifying the signal xi (step S2), and then release time control is performed (step S3). Here, in the configuration shown in FIG. 1, the time (attack time) from when a signal larger than the threshold value is input until the operation of the expander is started, and until the operation of the expander is stopped when the input signal becomes smaller than the threshold value. (Release time) can be set. In the release time control in step S3, control is performed to peak hold | xi | according to the set time.

Next, it is determined whether or not the region satisfies Li ≦ vth2 (step S4). If YES, the process proceeds to step S5 to calculate a control gain Gm = (vth2 / vth1) ^n-1, and then proceeds to step S10. . On the other hand, if NO, the process proceeds to step S6 to determine whether or not the region is Li ≧ vth1 (step S6). If YES, the process proceeds to step S7 to select Δ = 0, and then proceeds to step S9. Also, Li
If not ≧ vth1, that is, in the case of a region, Δ = vth
1-Li is selected, and the routine proceeds to step S9. Step S9
Then, Gm = 1−k (Δ + qΔ ² ) / Li is calculated. In step S10, the expanded gain Ge is set to 1, and
Next, attack time control is performed (step S11),
Then, yo = xi.Gm.Ge is calculated and output (step S12). In the attack time control, the gain Gm is integrated according to the set time.

According to such a configuration, when the expander of the algorithm shown in FIG. 1 is configured by the DSP, the number of processing steps is about 60 steps when the division accuracy in the division instruction is 24 bits (however, the attack time control And release time control processing are excluded).

FIG. 4 shows an upward expander characteristic of the acceleration saturation characteristic as a second embodiment, and FIG. 5 shows the processing. 4, in the region of the Gm = 1 Vth2 <Xi <Vth1 in the region of Xi ≦ Vth2, the Gm = 1-k (Δ + qΔ 2) / Li Xi ≧ Vth1 Gm in the region of ^{= (vth1 / vth2) n-} 1 Is controlled so that

In the region of Vth2 <Xi <Vth1, the acceleration saturation characteristic of the m-th order function that saturates the region at the threshold value Vth1 is obtained. Here, the equation (11) relating to the region in the conventional example is modified so that Gm = ｛( Li / vth1) ^n-1 ｝ · (vth1 / vth2)
Assuming that ⁿ⁻¹ , the expanded gain Ge can be obtained by using the equation (10) Gm = (vth1 / vth2) ⁿ⁻¹ relating to the region in the conventional example, and the DSP can be uniquely calculated if the threshold values Vth1 and Vth2 are given. And the processing steps do not increase.

The processing of the second embodiment will be described with reference to FIG. 5. First, similarly to the first embodiment, when an audio signal xi having positive and negative polarities is input (step S21). By rectifying the signal xi, Li = |
xi | is obtained (step S22), and then release time control is performed (step S23). Then Li ≦ vth2
(Step S24), the process proceeds to step S25 if YES, and proceeds to step S27 if NO. In step S25, the control gain Gm = 1 is set, and then the expand gain Ge = 1.
(Step S26), and the process proceeds to step S32.

In step S27, it is determined whether or not the region satisfies Li ≧ vth1, and if YES, the process proceeds to step S28 to select Δ = 0, and then proceeds to step S30. Also, L
When i ≧ vth1 is not satisfied, that is, in the case of a region, Δ = v
th1−Li is selected (step S29), and step S3 is selected.
Go to 0. In step S30, Gm = 1−k (Δ + qΔ)
² ) Calculate / Li and then expand gain Ge =
(Vth1 / vth2) ^n- 1 (step S31)
Proceed to step S32. At step S32, attack time control is performed, and then yo = xi.Gm.Ge is calculated and output (step S33).

Next, a third embodiment will be described.
FIG. 6 is an explanatory diagram showing an expanding process, a compressing process, and a limit process according to the third embodiment. FIG. 7 is a block diagram schematically showing an embodiment of an audio signal processing device according to the present invention. FIG. 9 is a block diagram functionally showing the configuration of the DSP of FIG. 7, FIG. 10 is an explanatory diagram showing an algorithm of the compressor of FIG. 9, and FIG. 11 is a diagram of FIG. FIG. 4 is an explanatory diagram illustrating an algorithm of a soft limiter.

In this device, as shown in FIG. 6, the expansion threshold values Vth1 and Vth2 are set with respect to the input level [dB].
, Compression threshold Vth-comp and limit threshold Vth-lim
it is given. However, Vth2 <Vth1 <Vth-comp <Vth-limit. Then, after performing the above-described accelerated saturation type downward expansion process, the compression threshold value Vth-c
Perform compression processing in the area above omp,
Next, limit processing is performed in an area equal to or larger than the limit threshold Vth-limit.

Next, the configuration of the audio signal processing device will be described with reference to FIG. The analog audio signal input via the input terminal 1 is input to the A / D converter 4 via the buffer 2 and the amplifier 3a or 3b and is converted into a digital audio signal. Then, the gain Gm of the digital audio signal is changed. Expand processing, compression processing, limit processing and the like are performed by the DSP 5 with the characteristics shown in FIG. The digital audio signal processed by the DSP 5 is converted into an analog audio signal by a D / A converter 6, and the analog audio signal is output via an analog switch (SW) 7 and an output terminal 8.

The CPU 10 controls the entire apparatus,
In particular, various parameters are set for the DSP 5. CP
U10 also includes a SW / encoder input unit 11 constituting an operation unit, an LED 12 and an LCD 1 constituting a display unit.
6 and the pattern control (PATTERN CON
TROL) I / O unit 13, a program ROM 14, a work RAM 15, and the like.

The SW / encoder input unit 11 sends various characteristic parameters as shown in FIG. 8 to the LPF 2 shown in FIG.
1, HPF 22, noise gate 23, exciter 24,
Parametric equalizer (PEQ1 to PEQ3) 2
5, the expander 26, the compressor 27, the side chain equalizer (EQ) 28, the output gain control unit 29, and the soft limiter 30 can be set. For example, after a signal larger than a threshold is input, the expander 26 and the compressor 27 can be set. And the time until the operation of the expander 26 and the compressor 27 is stopped (release time) when the input signal becomes smaller than the threshold value is settable.

Next, the DSP 5 will be described with reference to FIG. First, the LPF 21 and the HPF 22 perform processing to increase the intelligibility of the voice by cutting unnecessary low and high frequencies in the voice band, respectively. Next, the noise gate 23 performs the digital mute processing when the signal below the threshold value is continuous. By doing so, the S / N ratio is improved. Next, the exciter 24 performs a process of increasing the intelligibility of the sound of “sa line” by adding even harmonic components in the middle and high frequencies, and the three-stage parameter equalizer 25 corrects the frequency characteristics to thereby improve the intelligibility of the sound. Is performed.

The following expander 26 can use almost the same algorithm as that shown in FIG. 1 in order to realize the downward expand characteristic of the acceleration saturation characteristic shown in FIG. However, the second expanded gain Ge
Gm = constant value <1 in the region of Xi ≦ Vth2, Gm = constant value> 1 in the region of Xi ≧ Vth1, and the threshold Vth1 in the region of Vth2 <Xi <Vth1.
, The gain Gm of the acceleration saturation characteristic of the m-th order function that saturates with the gain Gm of the area is set. According to such an expansion characteristic, since the area where Gm> 1 is wide, the whisper can be clarified, and the S / N can be improved in the area where Gm <1 (background noise area). It can be a noise gate characteristic.

The compressor 27 performs a compression process of multiplying (MPY in the figure) by a level control gain Gm that increases at an accelerated rate by using an algorithm as shown in FIG. The approximation processing of the acceleration rise characteristic is given by the following equation when expressed by a quadratic function, similarly to the expander. Li> vth
In the region of -comp, Yo = Xi (t−d) × ｛1−k (Δ + qΔ ² ) / Li
According to such a compression process, since the level control gain Gm increases at an accelerated rate, it is possible to suppress the output sound volume in a loud voice region and prevent the sound from being inflected.

The output gain is controlled by the output gain control unit 29, and the soft limiter 30 is used to limit the impulsive sound, such as when the microphone is dropped, instantaneously without a time lag, in the region of Li> vth-limit. As shown in FIG. 11, limit processing for level control is performed so as to gradually saturate around vth-limit. Xi ≧ 0 Yo = Xi−k (Δ + qΔ ² ) Xi <0 Yo = Xi + k (Δ + qΔ ² ) According to such limit processing, the level control gain Gm is accelerated by the compression processing in the loud voice region. Even if it increases, it is possible to prevent the speaker from being damaged.

Although the above embodiment has been described as an audio signal processing apparatus, if a storage medium storing a program having each step of processing in this processing apparatus is loaded into a computer, the same processing as the above embodiment can be performed by using a general-purpose computer. An audio signal processing device can be realized. Specifically, by supplying this program from a recording medium on which this program is recorded to a computer and performing processing on the audio signal to be processed, it is possible to improve audio characteristics in a remote conference. it can. CP
As the speed of U increases, MMX (multimedia) -compatible computers can process signals with the CPU. Therefore, the processing in FIG. 1 can be realized by such a computer. That is, the processing of FIG. 1 is performed on the microphone input signal connected to the computer, so that clearer voice can be transmitted. If the present invention is implemented by a computer as described above, a special device is not required, so that audio processing can be easily realized. Therefore, the storage medium storing the program including the processing steps in the audio signal processing device of the present invention is a part of the present invention.

[0037]

As described above, according to the present invention, when the level of the input digital audio signal is in the range of the threshold value Vth1−Vth2 when the expanding process is performed, the threshold value Vth
Since the level control gain Gm approximated by an m-th order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the region of 1 or more, the output level change with respect to the input level near the high threshold Vth1 is obtained. The connection can be made natural by smoothing the sound. In addition, it is possible to improve the fall characteristic of the sound near the low threshold value Vth2 to prevent the sound from being cut off, and to reduce the number of signal processing steps when processing is performed using the DSP.

[Brief description of the drawings]

FIG. 1 is an explanatory diagram showing an algorithm of an expander which is a main part of an embodiment of an audio signal processing device according to the present invention.

FIG. 2 is a graph showing characteristics of the expander of FIG.

FIG. 3 is a flowchart for explaining processing of the expander in FIG. 1;

FIG. 4 is a graph showing characteristics of the expander according to the second embodiment.

FIG. 5 is a flowchart illustrating a process according to a second embodiment.

FIG. 6 is an explanatory diagram showing an expanding process, a compressing process, and a limit process according to a third embodiment.

FIG. 7 is a block diagram schematically showing an embodiment of an audio signal processing device according to the present invention.

FIG. 8 is an explanatory diagram showing setting parameters of the audio signal processing device of FIG. 7;

FIG. 9 is a block diagram functionally showing the configuration of the DSP of FIG. 7;

FIG. 10 is an explanatory diagram showing an algorithm of the compressor in FIG. 9;

FIG. 11 is an explanatory diagram showing an algorithm of the soft limiter of FIG. 9;

FIG. 12 is an explanatory diagram showing a conventional downward expanding characteristic.

FIG. 13 is an explanatory diagram showing a conventional upward expanding characteristic.

[Explanation of symbols]

 26 Expander (expanding means) 27 Compressor (compressing means) 30 Soft limiter (limit means)

Claims (8)

[Claims] 1. An audio signal processing apparatus for expanding a digital audio signal by multiplying the digital audio signal by a level control gain Gm, wherein threshold values for determining the level of an input digital audio signal are set to Vth1 and Vth2 (Vth1> Vth2). Assuming that the ratio is n, Gm = 1 in an area where the level of the input digital audio signal is equal to or higher than the threshold Vth1, and Gm = (Vth2 / Vth1) ^{n-1 in} an area where the level of the input digital audio signal is equal to or lower than the threshold Vth2. Audio signal level is threshold Vth1-V
Expanding means for multiplying the digital audio signal by a level control gain Gm approximated by an m-order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the area above the threshold Vth1 in the area of th2 An audio signal processing device characterized by the above-mentioned. 2. An audio signal processing apparatus for expanding a digital audio signal by multiplying the digital audio signal by a level control gain Gm, wherein threshold values for determining the level of an input digital audio signal are Vth1 and Vth2 (Vth1> Vth2). Assuming that the ratio is n, Gm = (Vth1 / Vth2) ^{n-1 in} an area where the level of the input digital audio signal is equal to or higher than the threshold value Vth1, Gm = 1 in an area where the level of the input digital audio signal is equal to or lower than the threshold value Vth2. Audio signal level is threshold Vth1-V
Expanding means for multiplying the digital audio signal by a level control gain Gm approximated by an m-order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the area above the threshold Vth1 in the area of th2 An audio signal processing device characterized by the above-mentioned. 3. A threshold value Vth-comp among digital audio signals expanded by said expanding means.
(> Vth1) or more digital audio signal,
3. The audio signal processing device according to claim 1, further comprising a compression means for multiplying a level control gain Gm whose value increases at an accelerating rate. 4. The digital audio signal compressed by the compressing means, wherein a threshold Vth-limit (>
Vth-comp)
4. The audio signal processing device according to claim 3, further comprising limit means for performing level control so that the value gradually becomes saturated near the threshold value Vth-limit. 5. For audio signal processing for expanding by multiplying a digital audio signal by a level control gain Gm, threshold values for determining the level of an input digital audio signal are Vth1 and Vth2 (Vth1> Vth2). Assuming that the expand ratio is n, Gm = 1 in an area where the level of the input digital audio signal is equal to or higher than the threshold Vth1, and Gm = (Vth2 / Vth1) ^{n-1 in} an area where the level of the input digital audio signal is equal to or lower than the threshold Vth2. Digital audio signal level is threshold Vth1-V
In the th2 region, an audio signal having an expanding step of multiplying the digital audio signal by a level control gain Gm approximated by an m-order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the region above the threshold value Vth1 A storage medium on which a processing program is recorded. 6. A threshold value for judging the level of an input digital audio signal for audio signal processing for expanding by multiplying a digital audio signal by a level control gain Gm is set to Vth1, Vth2 (Vth1> Vth2). Assuming that the expand ratio is n, Gm = (Vth1 / Vth2) ^{n-1 in} an area where the level of the input digital audio signal is equal to or higher than the threshold Vth1, Gm = 1 in an area where the level of the input digital audio signal is equal to or lower than the threshold Vth2, Digital audio signal level is threshold Vth1-V
In the th2 region, an audio signal having an expanding step of multiplying the digital audio signal by a level control gain Gm approximated by an m-order function (m is an integer of 2 or more) that saturates with the level control gain Gm in the region above the threshold value Vth1 A storage medium on which a processing program is recorded. 7. A threshold value Vth-c among digital audio signals expanded in the expanding step.
7. A compression step of multiplying a digital audio signal equal to or greater than omp (> Vth1) by a level control gain Gm whose value increases at an accelerating rate.
A storage medium on which the described audio signal processing program is recorded. 8. A threshold value Vth-limit among digital audio signals compressed in the compression step.
8. The audio signal processing program according to claim 7, further comprising a limit step of level-controlling the digital audio signal of (> Vth-comp) or more so that the value gradually becomes saturated near the threshold value Vth-limit. Storage medium.