JP3732681B2 - Audio sound canceling device - Google Patents

Description

[0001]
BACKGROUND OF THE INVENTION
The present invention relates to an apparatus for canceling audio sound by adaptive signal processing, and in particular, controls the gain of an amplifier that amplifies a reference signal by monitoring the coefficient value of the adaptive filter, and sets the level ratio between the reference signal and the target response to an optimum value. The present invention relates to an audio sound canceling apparatus that is automatically set to “.
[0002]
[Prior art]
When a speaker voice signal is input to a voice recognition device, a hands-free telephone, etc., input of a voice signal having a large S / N ratio with reduced audio sound and ambient noise improves the voice recognition rate or is comfortable. Necessary for proper communication. For this reason, various systems for reducing noise and audio sound have been proposed.
FIG. 3 is a block diagram of an apparatus for canceling an audio sound, where 1 is an audio source, 2 is an audio signal input, and a speaker that radiates the audio sound into the acoustic space. 3 is an audio sound radiated into the acoustic space. The microphones 4 perform adaptive signal processing so that the power of the error signal e is minimized, and the adaptive signal processing units 5 and 6 update the coefficient W of the adaptive filter. And an amplifier 7 provided between the adaptive signal processing unit 4 and a subtractor 7 which calculates the difference between the microphone output signal and the adaptive filter output and outputs an error signal e.
[0003]
Although not shown, the adaptive signal processing unit 4 includes an LMS calculation unit and an adaptive filter having an FIR digital filter configuration. The adaptive signal processing unit 4 uses the audio signal as the reference signal and the signal output from the microphone as the target signal, and updates the coefficient W of the adaptive filter so that the power of the error signal e is minimized, thereby suppressing the audio sound. I do. That is, by the adaptive signal processing, the adaptive filter comes to simulate the propagation characteristics from the speaker 2 to the microphone 3, and the audio sound is canceled.
[0004]
[Problems to be solved by the invention]
In the audio cancellation apparatus of FIG. 3, in order to increase the audio sound suppression performance, it is necessary to increase the accuracy of the coefficient W set in the adaptive filter. For this purpose, the amplitude ratio between the reference signal and the target response needs to be an appropriate value. However, in general, the amplitude of the target response varies depending on the positional relationship between the speaker 2 and the microphone 3. For this reason, when the acoustic space is a vehicle compartment, the gains of the amplifiers 4 and 5 cannot be set so that the amplitude ratio between the reference signal and the target response becomes an appropriate value unless the vehicle type on which the device is mounted is determined. As described above, conventionally, after the apparatus is mounted on a vehicle, the gains of the amplifiers 4 and 5 are adjusted to adjust the filter coefficient to an appropriate value. However, this adjustment work is troublesome and has a problem that the burden on the user is great.
As described above, an object of the present invention is to set the amplitude ratio of the reference signal and the target response so that the filter coefficient automatically becomes an appropriate value.
[0005]
[Means for Solving the Problems]
According to the present invention, an audio source that outputs an audio signal, a speaker that receives an audio signal and radiates audio sound to the acoustic space, a microphone that detects audio sound radiated to the acoustic space, and the audio signal The target signal is the signal and the signal output from the microphone, adaptive signal processing is performed so that the power of the error signal is minimized, the coefficient of the adaptive filter is updated, and the transfer characteristic from the speaker to the microphone is simulated by the adaptive filter An adaptive signal processing unit that is provided between the audio source and the adaptive signal processing unit, a variable gain amplifier that amplifies the audio signal, and a maximum value of the adaptive filter coefficient so that transfer characteristics from the speaker to the microphone can be correctly simulated. Set, monitor the coefficients of the adaptive filter, and Maximum coefficient value of filter is achieved by an audio sound canceling device having a gain setting section for controlling the gain of the amplifier so that the set value.
[0006]
DETAILED DESCRIPTION OF THE INVENTION
FIG. 1 is a block diagram of an audio sound canceling apparatus according to the present invention.
In the figure, 11 is an audio source, 12 is an audio signal input, a speaker that radiates audio sound into the acoustic space, 13 is a microphone that detects audio sound radiated into the acoustic space, and 14 is the power of the error signal e is minimum. An adaptive signal processing unit that performs adaptive signal processing so that the coefficient W of the adaptive filter is updated, 15 is a gain-fixed amplifier provided between the audio source 11 and the speaker 12, and 16 is the audio source 11 and the adaptive signal. A variable gain amplifier provided between the processing units 14, a subtractor 17 that calculates a difference between a microphone output as a target signal and an adaptive filter output and outputs an error signal e, and 18 a gain of the variable gain amplifier 16. It is a gain setting part to set.
[0007]
Although not shown, the adaptive signal processing unit 14 includes an LMS calculation unit and an adaptive filter having an FIR type digital filter configuration. The adaptive signal processing unit 14 uses the audio signal as the reference signal and the signal output from the microphone as the target signal, and updates the adaptive filter coefficient W so that the power of the error signal e is minimized.
[0008]
The gain setting unit 18 sets the gain of the amplifier 16 so that the adaptive filter coefficient W becomes an appropriate value. The absolute value of the filter coefficient W can take a value from 0 to 1, and the larger the coefficient coefficient, the higher the accuracy of the coefficient. However, when the coefficient becomes larger than 1 in the calculation of adaptive signal processing, it is clamped to 1, and the adaptive filter cannot accurately simulate the propagation characteristic from the speaker to the microphone. On the other hand, if the filter coefficient is too small, the accuracy of the coefficient is lowered, and similarly, the adaptive filter cannot accurately simulate the propagation characteristic from the speaker to the microphone.
Therefore, the gain setting unit 18 monitors the coefficient W of the adaptive filter and controls the gain of the amplifier 16 so that the maximum value of the filter coefficient becomes a set value, for example, 0.95. That is, the gain is controlled so that the filter coefficient increases if the maximum value of the filter coefficient W is smaller than 0.95, and the gain is controlled so that the filter coefficient decreases if the maximum value of the filter coefficient is larger than 0.95.
[0009]
When the reference signal is smaller than the target signal (reference signal <target signal), the filter coefficient W is increased by adaptive signal processing, and the reference signal = target signal. On the contrary, if the reference signal is larger than the target signal (reference signal> target signal), the filter coefficient W is decreased by the adaptive signal processing, and the reference signal = target signal. Therefore,
(1) The filter coefficient is smaller than 0.95, and in order to increase the filter coefficient, the gain of the amplifier 16 may be decreased so that the reference signal <target signal,
(2) In order to make the filter coefficient larger than 0.95 and reduce the filter coefficient, the gain of the amplifier 16 may be increased so that the reference signal> the target signal.
The gain setting unit 18 controls the gain of the amplifier 16 in accordance with the above (1) and (2), so that the maximum value of the filter coefficient becomes a set value, for example, 0.95.
[0010]
FIG. 2 is a processing flow of gain setting control of the present invention.
The adaptive signal processing unit 14 starts updating the filter coefficient W by adaptive signal processing (start learning, step 101), and stops updating the filter coefficient when the filter coefficient W converges to a certain value (learning stop, step 101). 102). The gain setting unit 18 reads the filter coefficient (step 103), obtains the maximum filter coefficient value, and checks whether the maximum filter coefficient is equal to the set value (= 0.95) (step 104). Exit. If the maximum filter coefficient is larger than the set value (= 0.95), the gain of the amplifier 16 is increased. If the maximum filter coefficient is smaller than the set value (= 0.95), the gain of the amplifier 16 is decreased (step 105). Thereafter, the process returns to step 101, the update of the filter coefficient is resumed, and the above processing is repeated until the maximum filter coefficient = the set value (0.95).
[0011]
In the above, the set value is 0.95, but the present invention is not limited to this value. Further, an appropriate range of filter coefficients, for example, 0.90 to 0.95 can be set, and the gain can be controlled so as to fall within this range.
The present invention has been described with reference to the embodiments. However, the present invention can be variously modified in accordance with the gist of the present invention described in the claims, and the present invention does not exclude these.
[0012]
【The invention's effect】
As described above, according to the present invention, since the accuracy of the filter coefficient can be increased, the performance of the system can be sufficiently exhibited. In addition, since these settings are performed automatically, it is possible to significantly reduce the load on the user when performing these settings compared to the conventional case.
[Brief description of the drawings]
FIG. 1 is a configuration diagram of an audio sound canceling apparatus according to the present invention.
FIG. 2 is a gain setting process flow of the present invention.
FIG. 3 is a configuration diagram of a conventional audio sound canceling apparatus.
[Explanation of symbols]
11. Audio source 12 Speaker 13 Microphone 14 Adaptive signal processing unit 15 Gain fixed amplifier 16 Gain variable amplifier 17 Subtractor 18 Gain setting unit

Claims (1)

In a device that cancels audio sound,
An audio source that outputs audio signals,
A speaker that receives an audio signal and radiates an audio sound into an acoustic space;
A microphone that detects audio sound radiated into the acoustic space,
The audio signal is the reference signal, the signal output from the microphone is the target signal, adaptive signal processing is performed so that the power of the error signal is minimized, the coefficient of the adaptive filter is updated, and the adaptive filter processes the signals from the speaker to the microphone. An adaptive signal processor that simulates the transfer characteristics ;
A variable gain amplifier that is provided between the audio source and the adaptive signal processing unit and amplifies the audio signal,
The maximum value of the adaptive filter coefficient is set so that the transfer characteristic from the speaker to the microphone can be correctly simulated, the coefficient of the adaptive filter is monitored, and the maximum coefficient value of the adaptive filter becomes the set value. A gain setting unit for controlling the gain of the amplifier;
An audio sound canceling device comprising:

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