JP2001024459A - Audio device - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide an audio device for listening to a music even in a noisy situation in the same way as a situation that there is not any noise as much as possible. SOLUTION: A gain/signal characteristics in various kinds of noise levels are preliminarily stored in a memory MM of a loudness compensation gain calculating part 8. A noise signal is separated from the output of a microphone 7 by a signal separating part 6, and inputted to the loudness compensation gain calculating part 8 with a music signal. The loudness compensation gain calculating part 8 wavelet converts those signals, and detects the level for each frequency band, and calculates the optimal gain for each frequency band by referring to the gain/signal characteristics stored in the memory MM, and sets it at a filter 3 for correction. The filter 3 for correction adds the gain to the music signal, and supplies it to a loudspeaker 5.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

The present invention relates to an audio device for controlling the loudness of an audio signal in a noisy environment to be equal to the loudness of an audio signal in a noise-free environment.

[0002]

2. Description of the Related Art In the case of a system for listening to music in a noisy environment such as car audio or the like, noise causes a hearing masking phenomenon (a phenomenon that hinders hearing) and makes it difficult to hear the music. To cope with such a problem, various devices such as an automatic volume device and a loudness correction device have been conventionally proposed.

FIG. 10 is a block diagram showing an example of an automatic volume control device. 16 is a variable gain control section to which an audio signal is input, 17 is an amplifier, and 18 is an audio signal corresponding to the audio signal emitted to an acoustic space. A speaker 19 for detecting a synthesized sound signal of audio sound and noise at a predetermined observation point in the acoustic space;
0 is a detector for calculating the average level of the audio signal, 2
Reference numeral 1 denotes a detection unit that calculates the average level of the synthesized sound signal (microphone signal) detected by the microphone, and 22 denotes a gain determination unit that determines the gain based on the difference between the average levels of the microphone signal and the audio signal. The detectors 20 and 21 calculate the average levels of the audio signal and the microphone signal, respectively. The calculator 23 subtracts the average level of the audio signal from the average level of the microphone signal, and the gain determiner 22 regards the difference as an estimated noise average level. The gain control unit 16 multiplies the audio signal by the determined gain and outputs the result. As a result, when the noise increases, the gain increases and the volume increases, and when the noise decreases, the gain decreases and the volume decreases.

[0004] The principle of calculating the noise average level estimation value in the automatic volume device is as follows. Assuming that the audio signal is S and the noise signal is N, the signal M captured by the microphone 19 is a superposition of the audio signal S and the noise signal N, so that M = S + N. Therefore, the signal average power E [M ² ] captured by the microphone is Becomes Here, E [•] is an expected value operator, which is equivalent to taking a long-term average. By subtracting the average power of the audio signal from the average power of the signal captured by the microphone, E [M ² ] −E [S ² ] = E [S ² ] + 2E [S · N] + E [N ² ] −E [ S ^2] = the 2E [S · N] + E [N 2]. Here, since the audio signal S and the noise signal N are uncorrelated, E [S
[N] approaches 0, and the average power of the noise signal N is obtained. As described above, long-term averaging is required to improve the calculation accuracy of the noise average level estimation value.

FIG. 11 is a block diagram showing an example of a loudness correction device. Reference numeral 24 denotes a variable gain control unit to which an audio signal is input, 25 denotes an amplifier, and 26 emits an audio sound corresponding to the audio signal to an acoustic space. The speaker 27 is a gain determining unit that determines the gain of each frequency band according to the loudness level curve based on the level of the input audio signal.

The unit of "sound volume (loudness)" perceived by a human is sone, and the volume of a pure sound of 1 kHz and 40 dB is defined as 1 sone. Since it is based on human perception, 2sone sounds twice as large as 1sone.
Loudness varies not only with sound intensity but also with the frequency spectrum. FIG. 12 is a graph obtained by connecting sound pressure levels of pure tones having the same loudness as a 1 kHz pure tone in a state where there is no external noise, and is called an equal loudness level curve. That is, the equal loudness level curve is a plot of the level of another frequency at which a person can hear the same magnitude as a 1 kHz sine wave. The equal loudness level curve indicates that as the level becomes lower, the sound in the low frequency range and the high frequency range will be heard lower than the sound in the intermediate frequency range or the sound will not be heard unless the level is raised. In the loudness correction device of FIG. 11, the gain determination unit 27 determines the gain for each frequency based on the equal loudness level curve so that the same audio signal level can be heard at the same level, and the gain control unit 24 determines the gain. Level adjustment is performed for each frequency based on the frequency.

[0007]

In the automatic volume control device as shown in FIG. 10, since the gain of all frequencies is uniformly adjusted, noises whose noise power is biased to a low frequency such as an automobile noise can be reduced. There is a problem that a low frequency component of music cannot be heard, and a high frequency component can be strongly heard. In addition, since the average level of noise is calculated uniformly in the entire frequency band, if the level calculation time is set in accordance with a low frequency in which the time variation of the noise signal is small, it is not possible to follow a high frequency in which time variation is severe, Conversely, when the level calculation time is set in accordance with the high frequency, there is a problem that the calculation error of the low frequency level increases.

Further, the loudness correction device shown in FIG. 11 does not take into account the influence of noise, and cannot respond to changes in noise. Furthermore, since the focus is only on the music signal input to the apparatus, there is a problem that the amount of correction is not optimal due to changes in the audio system or the vehicle. As described above, an object of the present invention is to provide an audio device capable of listening to music and the like as much as possible without noise even in the presence of noise.

[0009]

SUMMARY OF THE INVENTION An object of the present invention is to provide an audio apparatus for controlling the loudness of an audio signal in accordance with noise in an acoustic space, wherein the synthesized sound of the audio sound and the noise detected at a predetermined observation point in the acoustic space. A noise separating unit that separates a noise signal from a signal, and a gain / signal level characteristic in which a gain corresponding to the audio signal level is set so that the loudness of the audio signal in a noisy environment is equal to the loudness of the audio signal in a noise-free environment. A gain / signal level characteristic storage unit for storing a noise level for each noise level, a noise level calculation unit for calculating a noise level in each frequency band of the noise signal output from the noise separation unit by wavelet transform, and a frequency for each of the audio signals. Audio signal level in band by wavelet transform A signal level calculation unit to be output, and the gain / signal level characteristic according to the noise level of each frequency band calculated by the signal level calculation unit is extracted from the gain / signal level characteristic storage unit, and the A gain determining unit that determines a gain according to an audio signal level, a signal gain control unit that applies a gain of each frequency band determined by the gain determining unit to each frequency band component of the audio signal, and a signal gain control unit. The problem is solved by an audio device having a speaker that emits an audio sound corresponding to an output audio signal to an acoustic space.

[0010] In addition to the above configuration, a measuring unit for measuring an impulse response from the speaker to the observation point;
A first filter configured to simulate an acoustic system by setting a coefficient corresponding to an impulse response measured by the measuring unit, to provide an input audio signal with acoustic system characteristics, and to output the same characteristics as the signal gain control unit; And a second filter for correcting an output from the first filter. In this case, the noise separation unit separates the noise signal by calculating the synthesized sound signal and the output of the second filter.

The signal gain control unit may be, for example, a wavelet transform unit that performs wavelet transform on an audio signal and separates the audio signal into frequency bands, and a gain multiplication that multiplies each frequency band component of the audio signal by the gain of each frequency band. And an inverse wavelet transform unit that performs an inverse wavelet transform on the output of each gain multiplication unit.

[0012]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS The principle of the present invention will be described below, and then the embodiments of the present invention will be described. (A) Principle FIG. 1 shows the physical sound pressure level and the "loudness (called loudness)" of the sound that a human perceives when listening to it.
This is called a loudness curve. In the loudness curve, the horizontal axis is the physical sound pressure level (unit is Sound Pressure Level SPL (dB))
The vertical axis is the loudness (unit: sone) that quantifies the loudness of the sound felt by a person. In FIG. 1, (a) is a loudness curve in a quiet environment, and (b) is a loudness curve under noise. However, (b) indicates that the minimum audible value of a person is about 35.
This is a curve in noise that increases in dB, and this curve changes variously as the noise changes.

The loudness curve indicates that if the numerical value of the loudness on the vertical axis is the same, a person perceives that the sound has the same volume. Therefore, the sound that a person perceives as a volume of 0.1sone may be a physical sound pressure level of 12 dB SPL in the quiet environment of (a), but 37 dB S under the noise of (b).
A PL physical sound pressure level is required. In other words, if the sound of 12 dB SPL is output from the audio system in a quiet environment, the sound of the same volume may be felt under the noise of (b) unless the sound of 37 dB SPL is output from the audio system. Can not. In other words, in order to hear a sound that feels as large as 0.1sone under noise, a 25 dB gain must be added as compared with a case where the sound is heard in a quiet environment. In addition, the sound that a person perceives as one sound volume is a physical sound pressure level of 42 dB SPL in the quiet environment of (a), but requires a physical sound pressure level of 49 dB SPL under the noise of (b). Therefore, a gain of 7 dB must be added.

Thus, it is necessary to change the gain depending on the level of the audio signal even under the same noise.
FIG. 2 (solid line) shows a graph showing the relationship between the level of the audio signal and the gain under the same noise. The horizontal axis of FIG. 2 is the physical sound pressure level (corresponding to the sound signal level) in a quiet environment, and the vertical axis is the same size as when the audio sound was heard in a quiet environment under the noise of FIG. 1 (b). This is the gain value needed to sound audible. As shown in this figure, in order to hear the sound in a quiet environment under the noise, it is necessary to change the gain according to the physical sound pressure SPL. The broken line in the figure is a straight line approximation of the gain / signal level characteristic curve shown by the solid line.

In the system of the present invention, the relationship between the level and the gain of an audio signal (hereinafter, referred to as a gain table) as shown in FIG. 2 under various noises is checked beforehand and recorded in a memory or the like. deep. Then, by examining the noise in the passenger compartment, selecting a gain table according to the state, calculating the optimum gain from the level of the audio signal using the gain table, and adding the gain to the audio signal, the Even in this case, music and the like can be heard with the same size (perceived volume) as in a quiet environment.

In the present invention, a wavelet transform is used to calculate the noise level and the audio sound level. The wavelet transform divides noise and audio sound into frequency bands according to human auditory characteristics, and calculates a noise level and an audio sound level of each frequency band. Thus, the level of the audio signal can be appropriately corrected for each frequency band. In addition,
Details of the wavelet transform will be described later.

(B) Configuration FIG. 3 is a configuration diagram of an audio device according to an embodiment of the present invention. Note that this embodiment shows an example in which the present invention is applied to an in-vehicle audio device. In the figure, 1 is an audio source such as a CD player, 2 is a volume for adjusting the volume, 3 is an audio signal correction filter for adding an optimum gain for each audio signal band, 4 is an amplifier for amplifying the audio signal, and 5 is an amplifier for amplifying the audio signal. A speaker 7 for radiating an audio signal output from the amplifier 4 to an acoustic space (cabin);
Is a microphone that detects a synthesized sound of an audio sound and noise at a predetermined position (observation point) in an acoustic space and converts the synthesized sound into an electric signal (synthesized sound signal). Details of the filter 3 will be described later.

Reference numeral 6 denotes a signal separation unit that separates and outputs a noise signal from a synthesized sound signal output from the microphone 7 and adds a propagation characteristic (transfer characteristic) in an acoustic space to the audio signal and outputs the audio signal. Reference numeral 8 denotes a loudness compensation gain calculator that receives a noise signal and an audio signal (music signal), calculates an optimum gain for each frequency band, and rewrites a coefficient (gain) of the filter 3. The gain table for each noise level as shown in FIG. 2 is recorded in the memory MM in the loudness compensation gain calculator 8.
Note that, instead of the gain / signal level characteristic curve, a linear approximation formula shown by a broken line in FIG. 2 may be stored. The details of the loudness compensation gain calculator 8 will be described later.

Reference numeral 9 denotes a white noise source that generates white noise to identify the impulse response of the acoustic system C in the vehicle cabin. Reference numeral 10 denotes a switch that inputs the white noise to the signal separation unit 6 when identifying the impulse response in the acoustic space. . 11 is a mixer,
At the time of identifying an impulse response to be described later, the white noise signal output from the white noise source 9 is transmitted to the next-stage amplifier 4, and during normal operation, the audio signal input from the correction filter 3 is transmitted to the next-stage amplifier 4.

In the signal separating section 6, reference numeral 6a denotes a filter simulating the impulse response of the acoustic system C from the speaker 5 to the microphone 7, and 6b denotes a filter having the same configuration as the correction filter 3. Each time the coefficient changes, the coefficient of the filter 6b is also rewritten. An adder 6c subtracts the output of the filter 6b from the output of the microphone 7 to extract only a noise component. This is because systems based on the principles of the present invention need to separate the noise and music signal components and know their respective levels. Reference numeral 6d denotes an adaptive control device for identifying an impulse response of the acoustic system C, which is constituted by an adaptive control unit 61 and an adaptive filter 62. The adaptive control device 6d identifies an impulse response (transfer characteristic) C ＾ of the acoustic system C by performing adaptive control using, for example, an LMS (Least Mean Square) adaptation algorithm, and calculates a coefficient corresponding to the impulse response C ＾ by a FIR (Finite). Impulse Response) is set in the adaptive filter 62 having a digital filter configuration. Reference numeral 6e denotes a mixer, which transmits a white noise signal input from the filter 6d to an adder 6c at the next stage when an impulse response is identified, and converts an audio signal input from the filter 6b during a normal operation, as described later. To the adder 6c.

(C) Identification of the impulse response of the acoustic system C Before starting the normal operation, it is necessary to identify the impulse response of the acoustic system C and set it in the filter 6b. FIG. 4 is an explanatory diagram of the identification of the impulse response of the acoustic system C, and a signal path related to the identification of the impulse response is indicated by a thick line. When identifying the impulse response of the sound system C, the audio source 1 is turned off to stop outputting the audio signal.
As a result, the audio signal correction filters 3 and 6b and the loudness compensation gain calculator 8 have no influence on the identification control.

When the audio signal is stopped, a white noise signal is generated by the white noise source 9 and the switch 10 is turned on. As a result, the white noise signal reaches the speaker 5 through the amplifier 4 and is radiated as sound into an acoustic space (vehicle compartment). The white noise emitted into the acoustic space is detected by the microphone 7 together with the surrounding noise, and the microphone 7 outputs a synthesized sound signal. Further, the white noise signal output from the white noise source 9 is input to the adaptive control device 6d, and is subjected to a filtering process by the adaptive filter 62. The arithmetic unit 6c subtracts the output signal of the adaptive filter 62 from the synthesized sound signal output from the microphone 7, and uses the difference as an error signal e to obtain an adaptive control device 6d.
Feedback to The adaptive control unit 61 performs adaptive control by the LMS adaptive algorithm so as to minimize the power of the error signal e, and sets the coefficient of the adaptive filter 62.
Thereafter, the above adaptive control is repeated, and finally the output of the adaptive filter 62 and the output of the microphone 7 become equal,
Thus, the adaptive filter 62 supplies the impulse response C 応 答 of the acoustic system C (strictly speaking, from the input of the amplifier 4 to the microphone 7
(Impulse response (transfer characteristic) up to).

After the identification of the impulse response of the acoustic system C is completed, the coefficient of the adaptive filter 62 is copied to the filter 6a, and the switch 10 is turned off to put it in a normal operable state.
Such identification control of the impulse response of the acoustic system C is performed when the user attaches the product to the vehicle. By performing this identification control, different amplifiers 4
Gain characteristics and the frequency characteristics of the acoustic system can be corrected.

(D) Normal operation In normal operation, the filter 6a adds an impulse response characteristic C ＾ of the acoustic system C to the audio signal input from the volume 2 to generate an audio signal at the observation point (microphone position). It is input to the loudness compensation gain calculator 8. The arithmetic unit 6c extracts a noise signal by subtracting the audio signal component from the synthesized sound signal output from the microphone 7, and inputs the noise signal to the loudness compensation gain calculation unit 8.

The loudness compensation gain calculating section 8 calculates the level of each frequency band of the noise signal when the noise signal is input, and calculates the level of each frequency band of the audio signal when the audio signal is input. calculate. Next, the loudness compensation gain calculator 8 selects a gain / signal level characteristic corresponding to the noise level from the memory MM, and refers to the gain / signal level characteristic to obtain a gain (coefficient) corresponding to the audio signal level for each frequency band. ) Is determined and set to the correction filter 3, and at the same time, the gain is copied to the filter 6b. That is, the loudness compensation gain calculator 8
Determines, for each frequency band, a gain for making the loudness of an audio signal in an environment where noise is generated equal to the loudness of an audio signal in an environment without noise, and sets the gain in the correction filter 3 and the filter 6b.

The correction filters 3 and 6b add the gain of each frequency band to each frequency band component of the input audio signal and output. Thereafter, the same control as described above is repeated, so that music such as car audio can be enjoyed as much as possible without noise even in the presence of noise. (E) Loudness Compensation Gain Calculation Unit The loudness compensation gain calculation unit 8 stores in advance the gain / signal level characteristics at various noise levels in the memory MM, and calculates the gain / signal level according to the actual noise level in the vehicle interior.
A signal level characteristic is selected, and an optimum gain according to the audio signal level is calculated for each frequency band with reference to the gain / signal level characteristic and output.

FIG. 5 is a block diagram of the loudness compensation gain calculating section 8. Reference numeral 81 denotes a first wavelet transform calculating section that outputs an average value of noise signals in each frequency band by wavelet transform, and 82 denotes each of the first and second wavelet transform arithmetic sections. A second wavelet transform operation unit 83 for outputting the average value of the audio signal in the frequency band is a noise level adjustment unit, which uses a well-known Zwicker loudness calculation method (ISO532B) or Stevens' loudness calculation method (ISO532A). The noise signal level of each frequency is adjusted in consideration of human auditory characteristics. 8
4 is a gain / signal level characteristic selection unit for selecting a gain / signal level characteristic according to a noise level for each frequency band;
Numeral 85 stores the gain / signal level characteristics (FIG. 2) at various noise levels in the memory MM, and refers to the gain / signal level characteristics according to the noise level to determine the optimum gain according to the audio signal level in the frequency band. This is a frequency band gain determination unit that outputs the frequency band gain every time.

The power spectrum of the noise is not flat,
Not all audio sounds are masked uniformly over the entire frequency band. That is, since the degree of masking varies depending on the noise level of each frequency band, it is necessary to calculate the gain for each frequency band and cope with it. In this case, noise of a certain frequency causes masking not only of audio sound of the same frequency but also audio sound of a higher frequency. For this reason,
As described above, the noise level adjustment unit 83 adjusts the noise signal level for each frequency band using the Zwicker's loudness calculation method and Stevens's loudness calculation method in consideration of human auditory characteristics.

Hereinafter, the wavelet transform will be described. The wavelet transform is defined by the following equation (2) while the short-time Fourier transform is defined by the following equation (1).

[0030]

(Equation 1)

Here, W (t−b) is a window function.

[0032]

(Equation 2)

In the equation (2), (Ψ (x−b) / a)
Is called a mother wavelet, and is a function as shown in FIG. 6, for example. In FIG. 6, a is referred to as a scale parameter, and is a parameter for expanding / contracting the mother wavelet by a factor. b is a parameter for translating the mother wavelet by b. Here, consider the result of frequency analysis performed by applying window functions having various time lengths to a sine wave as shown in FIG. The upper part of the figure is a time waveform, and the lower part is a corresponding frequency spectrum. As can be seen from this figure, when the value of a is taken so that the time window length becomes wider, the accuracy of the frequency analysis becomes better, and as the time window becomes narrower, the accuracy of the frequency analysis becomes worse. Conversely, if the accuracy of the frequency analysis is to be increased, the length of the time window must be increased, and the accuracy of the time analysis deteriorates.

The wavelet transform means that the time variation such as low frequency is small (in other words, the time analysis accuracy may be low), and the scale parameter a is set to be long where the frequency resolution is high in the human ear. Conversely, the frequency resolution is increased and the time resolution is decreased, and the time variation such as high frequency is large (in other words, high time analysis accuracy is required). In the human ear, the scale parameter a is shortened where the frequency resolution is low. In this case, the analysis is performed by increasing the time resolution and, conversely, by lowering the frequency resolution, so that an analysis closer to human auditory characteristics can be performed. On the other hand, in the short-time Fourier transform, the time resolution and the frequency resolution are constant at all frequencies.

FIG. 8 shows a comparison between the time resolution and the frequency resolution of the wavelet transform and the short-time Fourier transform. FIG. 2 also shows the case of the Fourier transform and the case of the time signal. 8, the horizontal axis represents time t, and the vertical axis represents angular frequency ω (frequency f = 1 / T = ω / 2π). That is, in the case of the Fourier transform, although the frequency resolution is high, there is no information on time. If the time waveform is used, the frequency cannot be analyzed. In the short-time Fourier transform, the frequency resolution and the time resolution are constant at all frequencies. On the other hand, in the wavelet transform, it is possible to increase the time resolution instead of decreasing the frequency resolution in a high frequency region, and to increase the frequency resolution instead of decreasing the time resolution in a low frequency region.
Thus, in the wavelet transform, the time resolution and the frequency resolution change according to the frequency.

(F) Correction Filter The compensation filter 3 adds a gain for each frequency band output from the loudness compensation gain calculator 8 to the audio signal. FIG. 9 is a configuration diagram of a correction filter.
31 is a wavelet transform operation unit, 32 is provided with a large number of variable gain amplifiers AMP1 to AMPn, and multiplies each frequency band component of the audio signal by the gain of each frequency band output from the loudness compensation gain calculation unit 8 to output a variable gain. And 33, an inverse wavelet transform operation unit that performs an inverse wavelet transform on the output signal of the variable gain amplifier.

The wavelet transform operation unit 31 performs a wavelet transform on the audio signal and separates and outputs the audio signal for each frequency band. The gain variable unit 32 multiplies the output signal of each frequency band by the gain of the corresponding frequency band output from the loudness compensation gain calculation unit 8. The inverse wavelet transform operation unit 33 synthesizes and outputs the signals of each frequency band whose gain has been adjusted by performing an inverse wavelet transform. As described above, gain control is performed so as to compensate for masking for each frequency band component of the input audio signal, and it is possible to make the noise-free state as close as possible even in a noise environment.

However, if the gain changes rapidly with time, the output waveform will be discontinuous. Therefore, it is preferable to gradually update the gain by controlling the gain according to the following equation: G (n) = αG (n−1) + βGm so as to prevent a sudden change in the gain. Here, in the above equation, G (n) is the time n
, G (n-1) is the gain characteristic at time (n-1), and Gm is the gain characteristic calculated by the loudness compensation gain calculator. Α and β are α +
This coefficient satisfies β = 1.

In the above embodiment, an example is shown in which the present invention is applied to an in-vehicle audio device, but the present invention is not limited to the in-vehicle audio device. INDUSTRIAL APPLICABILITY The present invention can be applied to a home audio device and other audio devices in addition to a vehicle-mounted audio device.

[0040]

As described above, according to the present invention,
Since the noise signal and the audio signal are subjected to the wavelet transform to detect the noise level and the audio signal level for each frequency band, the level of the audio signal for each frequency band corresponding to the noise is adjusted in a state in which human hearing characteristics are close. Can be. Further, according to the present invention, gain / signal level characteristics at various noise levels are stored in the gain / signal level characteristic storage unit in advance, and a gain / signal level characteristic according to actual noise in the acoustic space is selected. An appropriate gain corresponding to the audio signal level is calculated with reference to the gain / signal level characteristics, and the gain is added to the audio signal. Etc. can be enjoyed.

In the present invention, an acoustic impulse response from a speaker to an observation point is measured, a coefficient corresponding to the impulse response is set in a filter to simulate the acoustic system, and an audio signal is input to the filter. The output of the sound system is given by adding the characteristics of the sound system, and the filter output is subtracted from the synthesized sound signal detected at the observation point to output a noise signal. Even when the frequency characteristics change, it is possible to appropriately cope with the case.

[Brief description of the drawings]

FIG. 1 is an explanatory diagram of a loudness curve.

FIG. 2 is a gain-signal level characteristic diagram.

FIG. 3 is a configuration diagram of an audio device according to an embodiment of the present invention.

FIG. 4 is an explanatory diagram showing a method for identifying an impulse response in an acoustic space.

FIG. 5 is a configuration diagram of a loudness compensation gain calculation unit.

FIG. 6 is a diagram illustrating an example of a mother wavelet.

FIG. 7 is a diagram illustrating a result of frequency analysis performed by applying a window function to a sine wave.

FIG. 8 is a diagram showing a comparison of time-frequency resolution by Fourier transform, time waveform, short-time Fourier transform, and wavelet transform.

FIG. 9 is a configuration diagram of a correction filter.

FIG. 10 is a configuration diagram illustrating an example of an automatic volume device.

FIG. 11 is a configuration diagram illustrating an example of a loudness correction device.

FIG. 12 is a diagram showing a loudness level curve such as a pure tone and a minimum audible value in a free sound field.

[Explanation of symbols]

 Reference Signs List 1 audio source, 2 volume adjustment volume, 3 correction filter, 4 amplifier, 5 speaker, 6 signal separation unit, 7 microphone, 8 loudness compensation gain calculation unit, 9 white noise source, 10 switch, 31, 81, 82 wavelet Conversion operation unit, 32 gain variable unit, 33 inverse wavelet transform operation unit, 83 noise level adjustment unit, 84 gain / signal level characteristic selection unit, 85 frequency band gain determination unit.

Claims (3)

[Claims] 1. An audio apparatus for controlling a loudness of an audio signal in accordance with noise in an acoustic space, comprising: a noise separator for separating a noise signal from a synthesized sound signal of the audio sound and the noise detected at a predetermined observation point in the acoustic space. A gain and a gain for storing, for each noise level, a gain and a signal level characteristic corresponding to a gain corresponding to the audio signal level so that the loudness of the audio signal in a noisy environment is equal to the loudness of the audio signal in a no noise environment. A signal level characteristic storage unit, a noise level calculation unit that calculates a noise level in each frequency band of the noise signal output from the noise separation unit by wavelet transform, and an audio signal level in each frequency band of the audio signal by wavelet transform. A signal level calculator for calculating, The gain / signal level characteristic according to the noise level of each frequency band calculated by the signal level calculation unit is extracted from the gain / signal level characteristic storage unit, and the gain according to the audio signal level is calculated for each frequency band. A gain determiner for determining, a signal gain controller for applying a gain of each frequency band determined by the gain determiner to each frequency band component of the audio signal, and an audio corresponding to an audio signal output from the signal gain controller. An audio device comprising: a speaker that emits sound to an acoustic space. 2. A measuring section for measuring an impulse response from the speaker to the observation point, and a coefficient corresponding to the impulse response measured by the measuring section is set to simulate an acoustic system, and an acoustic system is applied to an input audio signal. And a second filter that has the same characteristics as the signal gain control unit and corrects the output from the first filter. The noise separation unit includes: 2. The audio device according to claim 1, wherein a noise signal is separated by an operation of the synthesized sound signal and an output of the second filter. 3. A signal gain control unit, comprising: a wavelet transform unit configured to perform a wavelet transform on an audio signal to separate each frequency band; and a gain multiplication unit that multiplies each frequency band component of the audio signal by a gain of each frequency band. 2. The audio device according to claim 1, wherein the audio device is configured by: a unit; and an inverse wavelet transform unit that performs an inverse wavelet transform on an output of each gain multiplication unit.