KR101182017B1 - Method and Apparatus for removing noise from signals inputted to a plurality of microphones in a portable terminal - Google Patents

Abstract

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to noise cancellation of signals in portable terminals, and more particularly, to a method and apparatus for removing noise from phased signals input to a portable terminal. In a mobile terminal having a first microphone and a second microphone of the present invention, a method for removing noise from signals input to the microphones includes first and second signals input to the first and second microphones. Converting the signals into the signals in the frequency domain, calculating phase values and frequency values for the signals in the first and second frequency domains, calculating the difference between the calculated phase values, Calculating a gain value inversely proportional to the difference between the phase values, and multiplying the gain value by the frequency-specific magnitude values of the signals in the first and second frequency domains to remove the noise in the first and second frequency domains. And converting the noise-free first and second frequency domain signals into time domain signals.

Phase difference, attenuation gain, energy ratio per channel, noise

Description

Method and Apparatus for removing noise from signals inputted to a plurality of microphones in a portable terminal

1 is a diagram illustrating a structure for removing noise using a beamforming scheme.

2A and 2B are exemplary views of a portable terminal having two microphones according to a preferred embodiment of the present invention.

3 illustrates a structure for reducing noise levels in accordance with a preferred embodiment of the present invention.

4 is a flow chart illustrating a noise reduction process according to a preferred embodiment of the present invention.

5 is a graph showing a signal received through two microphones and a signal without noise in accordance with a preferred embodiment of the present invention.

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to noise cancellation of signals in portable terminals, and more particularly, to a method and apparatus for removing noise from phased signals input to a portable terminal.

In the case of voice communication with the other party by using the mobile terminal, the mobile terminal is usually placed close to the user's face to make a voice call in a situation where the distance between the microphone and the mouth is close. At this time, since the distance between the microphone and the signal is close, the speaker's voice is input loudly and is relatively less affected by ambient noise. However, recently, in the case of a communication using a video call or voice recognition using a portable terminal, the portable terminal cannot be attached to a face as in the past. That is, in order to make a video call, a call must be made while looking at the other party's face displayed on the screen of the portable terminal, so that the distance between the microphone mounted on the portable terminal and the speaker's mouth becomes farther. In addition, in order to use the voice recognition application program while using the menu of the screen window of the mobile terminal, the distance between the microphone of the portable terminal and the speaker's mouth becomes far.

However, as the speaker's mouth, that is, the distance between the signal source and the microphone, increases, the reception sensitivity decreases in inverse proportion to the square of the distance between the signal source and the microphone, and ambient noise is inputted with energy similar to that of the voice signal. Therefore, the sound quality of the voice signal is deteriorated, which makes it difficult to transmit the content in the case of a video call, and causes a problem of recognition error in the case of communication using voice recognition. Therefore, to improve this, conventionally, a single channel noise cancellation method such as a spectral subtraction method, a Kalman filter, and a Wiener filter has been used. In order to have good noise canceling performance, we have found a way to use multiple microphone arrays.

The use of multiple microphones can create phase differences between input signals. When a signal is input through a plurality of microphones or the like, if the signal is input through the microphones at the same time, there is no phase difference between the signals input through the microphones, but the microphone is different from the microphones because the signal is different from the microphones. If they are not simultaneously input to the field, a phase difference occurs.

In this regard, a portable terminal having two microphones will be described in detail. If the speaker's voice signal comes in front of the two microphones, the two signals input to the two microphones have the same energy or phase. However, for signals coming from directions other than the front, a difference occurs between the two microphone input signals. That is, the input signal of the microphone that receives the signal first is larger in energy than the input signal of the other microphone. There is also a time delay between the two microphone input signals, depending on the microphone spacing and the angle of incidence of the signal. As such, there is no phase difference because there is no delay between the signals coming from the two microphones, but a phase difference occurs because there is a delay between signals coming from a direction other than the front side. Since the generated phase difference can be considered to be caused by noise, the phase difference information can be used as a measure of the presence and degree of noise. Therefore, since the noise is severe at a frequency where the phase difference occurs a lot, the noise canceling degree can be strengthened, and the noise is weak at the frequency where the phase difference is small, so the noise removal degree can be weakened to remove the noise.

As a noise canceling method using the plurality of microphone arrays, a beamforming method and an independent component analysis method may be used.

The independent component analysis method separates independent signals from the two signals when two voice or sound signals are mixed. Since the independent component analysis method follows a conventional operating principle, a detailed description thereof will be omitted. However, since the independent component analysis method requires various unrealistic assumptions, it is difficult to apply in a real environment in which various noises and reverberations exist. Therefore, since the independent component analysis method is sensitive to the direction of the sound source to be separated and the microphone direction setting, it is difficult to apply to signals coming from various directions. In the case of far-talking, noise cancellation performance is reduced due to a small difference in energy levels of input signals between two microphones, and a signal distortion becomes larger when signal separation fails.

The beamforming method has a structure as shown in FIG. 1.

1 is a diagram illustrating a structure for removing noise by using a beamforming method.

# MIC1 (110) is a microphone that receives a voice signal prior to # MIC2 (130), # MIC2 (130) is a microphone that receives a signal (eg, noise) other than the voice signal before # MIC1 (110) do.

When an analog signal is input through the # MIC2 130 which first receives a signal other than the voice signal, the analog signal is converted into a digital signal by an analog / digital converter (hereinafter referred to as 'A / D') 140 and output. do. The signal output from the A / D 140 is input to a delay 150 to delay the signal input time difference between the # MIC1 110 and # MIC2 130. Here, it is assumed that a signal is input to # MIC2 130 before # MIC1 110.

When an analog signal is input through the # MIC1 110, the A / D 120 is converted into a digital signal and output. The signal output from the A / D 120 is transferred to the summer 160. In this case, the delayed signal through the delayer 150 is also transmitted to the summer 160 and added or subtracted from the signal transmitted from the A / D 120.

The signal input through # MIC2 130 which receives a signal other than the voice signal first is likely to be unnecessary noise. Accordingly, the noise is finally '0' in the summer 160 by the beamforming method of FIG. 1.

As described above, the beamforming method is a robust algorithm capable of restoring the original signal without distortion, but has a limitation in improving the signal-to-noise ratio (SNR). It is difficult to obtain a clean original signal. In addition, the signal-to-noise ratio is improved as the number of microphones increases. However, there is a limitation in mounting a plurality of microphones in a small device such as a mobile terminal. The forming method is very limited to a portable terminal.

Therefore, there is a need for a method of efficiently removing noise in a signal in a noisy environment.

Accordingly, the present invention provides a method and apparatus for reducing noise levels in a noisy environment.

The present invention provides a method and apparatus for improving sound quality and recognition rate by using two omnidirectional microphones to reduce the noise level in a noisy environment to improve the signal-to-noise ratio and to minimize the sound distortion that may occur when the noise is removed. To provide.

In a mobile terminal having a first microphone and a second microphone according to the present invention, a method for removing noise from the signals input from the first microphone and the second microphone includes: receiving the input signals from the first and second frequencies; Converting the signals to the signals in the region, calculating phase values and frequency values for the signals in the first and second frequency domains, calculating a difference between the calculated phase values, Calculating a gain value inversely proportional to the difference between the phase values, and multiplying the gain value by the frequency-specific magnitude values of the signals in the first and second frequency domains to remove the noise in the first and second frequency domains. And converting the noise-free first and second frequency domain signals into time domain signals.

In addition, in a mobile terminal having a first microphone and a second microphone according to the present invention, an apparatus for removing noise from the signals input to the first microphone and the second microphone, the first and second input signals A frequency domain converter for converting signals in the frequency domain, a phase unit for calculating phase values for the signals in the first and second frequency domains, and a signal for the signals in the first and second frequency domains. A signal in the first and second frequency domains, the magnitude unit calculating a magnitude value for each frequency, a phase difference calculator calculating a difference between the calculated phase values, and a gain value inversely proportional to the difference between the phase values. And a gain calculator for calculating a value of the first and second frequency domains to remove noise by multiplying the gain values by frequency values of the signals of the first and second frequency domains, respectively. And a masking section for obtaining call and includes the noise removing the first and the time domain conversion unit for converting the output signals of the second frequency domain into a signal in the time domain.

The foregoing is a somewhat broad summary of features and technical advantages of the present invention in order that those skilled in the art will be better able to understand it from the following detailed description of the present invention. Additional features and advantages of the present invention, which form the subject matter of the claims of the invention, in addition to those features and advantages, will be better understood from the following detailed description of the invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS Hereinafter, a detailed description of preferred embodiments of the present invention will be described with reference to the accompanying drawings. Those skilled in the art will recognize that the disclosed concept and specific embodiments of the invention described below may be changed or modified to achieve the technical problem to be achieved by the present invention. Those skilled in the art will also recognize that concepts and structures equivalent to the concepts and structures disclosed herein do not depart from the spirit and scope of the broadest form of the invention. In the following description of the present invention, if it is determined that a detailed description of a related known function or configuration may unnecessarily obscure the subject matter of the present invention, the detailed description thereof will be omitted.

In order to explain the noise reduction method proposed by the present invention, it is assumed that two omni-directional microphones are applied among a plurality of microphone arrays. Here, the omnidirectional microphone refers to a microphone that receives the same signal from all directions without biasing any specific direction. As described above, when the signal is input through the two microphones, if the signal is input through the microphone at the same time, there is no phase difference between the signals input through the microphone, but the signal is different from the microphone If the microphones are not simultaneously input, a phase difference is generated.

In detail, if two microphones receive a voice signal from the front in an environment without ambient noise, there is no phase difference between the signals input to the two microphones. The phase difference (Phase_Diff) in this case is '0' as shown in Equation 1 below. One input of the two microphones is called a first microphone channel signal, and the other input is called a second microphone channel signal.

Here, CH1 (K) and CH2 (K) are frequency domain transformed signals of the first microphone channel signal and the second microphone channel signal, respectively, and K represents a frequency.

At this time, since the energy levels are almost the same, the magnitude ratio Mag_Ratio of the two microphone channel signals becomes 1 as shown in Equation 2 below.

However, if a signal other than the voice signal is input to the two microphones, since they are input at various angles from various directions, a phase difference occurs between the signals of the two microphone channels in this case. In this case, the phase difference may be caused by noise rather than by voice signals. Therefore, the present invention can lower the noise level by removing the phase difference signal, and can improve the sound quality during voice communication.

The phase difference is represented by the presence and level of noise at a specific frequency of a specific frame. Therefore, using the phase difference, the attenuation gain G (K) according to the frequency is obtained by <Equation 3>, and the energy ratio values MG1 (K) and MG2 (K) for each channel to the attenuation gain. ) To suppress noise. A large phase difference at that frequency indicates a loud noise, so multiply the attenuation gain close to '0' by the microphone channel signal to reduce the effect of noise. On the contrary, if the phase difference is small at the corresponding frequency, the noise is weak. Therefore, multiply the attenuation gain close to 1 by the microphone channel signal to reduce the loss of the original signal. Energy ratio values MG1 (K) and MG2 (K) for the respective channels may be obtained by the following Equations 4 and 5, respectively.

Here, in the equation for obtaining G (K), 1 of the denominator is for preventing division by 0, and C is a constant that determines the strength of the gain. If the C value is increased, the gain is strongly applied as a whole to distort the voice signal. On the contrary, if the C value is decreased, the gain is weakly applied, leaving much noise. This C value can be determined experimentally.

Noise may be removed from each microphone channel signal by using the attenuation gain value calculated from Equation 3, Equation 4, and Equation 5 and the energy ratio value for each microphone channel. According to Equation 6, noise can be removed by multiplying the attenuation gain value by the energy ratio value and multiplying the frequency-domain transformed microphone channel signal.

After the resulting signals of Equation 6 are transformed into signals in the time domain through time-domain transform (IFFT), either one channel is selected or an algorithm using two channels such as beamforming described above is applied. By combining, additional noise cancellation can be achieved.

2A and 2B are exemplary views of a portable terminal having two microphones according to a preferred embodiment of the present invention.

Referring to FIG. 2A, the mobile terminal 210 includes two microphones MIC1 220 and MIC2 230 at both sides of the front lower surface.

When two microphones are mounted on the front of the mobile terminal as shown in FIG. 2A, a delay is hardly generated between two microphone signals for a voice signal at a long distance and the magnitudes of frequencies are the same. Does not occur. Therefore, the phase difference between the two microphone signals is not caused by the voice signal but by the ambient noise. Therefore, when the microphones having the structure as shown in FIG. 2A are provided, the phase difference due to the ambient noise is calculated, the gain values for each channel are obtained, and the channel whose noise is suppressed by multiplying each channel signal by frequency domain conversion. Get signal.

Referring to FIG. 2B, the mobile terminal 250 includes a microphone (MIC1 260) at the front of the terminal and a microphone (MIC2 270) at the rear of the terminal.

As shown in FIG. 2B, when a microphone is mounted to the front and rear of the portable terminal, delay is generated between the voice signals inputted through the two microphones according to a sound path corresponding to the distance between the two microphones. Energy and phase differences occur. However, if the phase compensation is applied as much as this delay, the phase difference can be eliminated for the voice signal. Therefore, if the phase compensation for the difference according to the position of the microphone, the phase difference generated between the two microphone signals can be said to be due to the ambient noise. Therefore, when the microphones having the structure as shown in FIG. 2B are obtained, the gain values are obtained by using the phase difference and multiplied by the frequency-domain transformed channel signal to obtain a noise suppressed channel signal.

3 is a diagram illustrating a structure for reducing a noise level according to a preferred embodiment of the present invention.

Referring to FIG. 3, analog signals having a phase difference are input through the microphone 1 (# MIC1 302) and the microphone 2 (# MIC2 304).

The A / Ds 305 and 307 convert each of the input analog signals into digital signals and output the digital signals to the windows 310 and 312.

The windowing units 310 and 312 may window the transformed digital signals into a frame having a predetermined size for performing the frequency domain transformation, thereby performing a frequency domain transform unit (FFT). Output to (315, 317).

The frequency domain converters 315 and 317 convert the signal of the windowed time domain into a frequency domain signal and output the signals to the phase units 320 and 322 and the size units 325 and 327, respectively.

The phase units 320 and 322 calculate phase values from the signals input from the frequency domain converters 315 and 317, respectively, and output the phase values to the phase difference calculator 335.

The magnitude units 325 and 327 calculate magnitude values of respective frequencies from the signals input from the frequency domain converters 315 and 317 and output the calculated magnitude values to the energy ratio calculator 350.

Block 360 is a block for removing noise proposed in the present invention, the position compensation unit 330, phase difference calculation unit 335, attenuation gain calculation unit 340, masking unit (345, 347), energy ratio calculation unit It consists of 350.

As described above, the phase compensator 330 uses the distance difference between the microphones, that is, the delay value generated between the voice signals input according to the sound path, to the phase difference calculator 335 in order to make only a signal estimated as noise, not a voice signal. Output By doing so, it is possible to compensate for the phase difference that may occur in the speech signal. In other words, according to the phase difference between the signals input to the microphone 1 302 and the microphone 2 304, a delay value corresponding to a sound path between two microphones stored in advance is provided to the phase difference calculator 335. For example, if the microphones are mounted as shown in FIG. 2A, '0' is provided to the phase difference calculator 335 for phase compensation. However, if the microphones are mounted as shown in FIG. 2B, phase compensation is provided by providing a prestored delay value to the phase difference calculator 335. Using the delay value provided as described above, the phase difference calculator 335 calculates the phase difference by Equation 7 below. Accordingly, in order to obtain the phase difference from the phase difference calculator 335, the phase compensator 330 outputs a delay value as '0' to the phase difference calculator 335 in the case of FIG. 2A. In the case of FIG. 2B, a predetermined delay value is output to the phase difference calculator 335.

Here, Delay is the number of delay samples according to the distance between two microphones, that is, the delay value.

The phase difference calculator 335 calculates the phase difference using the phase values output from the phase units 320 and 322. Here, the phase difference may be different from that of the source signal being simultaneously input to the microphone 1 (# MIC1 302) and the microphone 2 (# MIC2 304), respectively, to the microphone 1 302 and the microphone 2 304. This means the difference in phase between the input signals. The phase difference value calculated by the phase difference calculator 335 is output to the attenuation gain calculator 340. The phase difference may be calculated by applying a delay value as '0' to Equation 7 when there is no delay between signals. In addition, when there is a delay between signals, the phase difference may be calculated by applying the delay value provided from the position compensator 330 to Equation (7).

The attenuation gain calculator 340 calculates the attenuation gain using the phase difference value output from the phase difference calculator 335 and outputs the attenuation gain to the masking units 345 and 347. The attenuation gain can be obtained by using Equation 3 described above.

The energy ratio calculator 350 calculates an energy ratio for each microphone channel using the magnitude values of the frequencies output from the magnitude units 325 and 327, and then masks the energy ratios 345 and 347. Will output The energy ratio for each of the microphones may be obtained by using Equations 4 and 5, respectively.

The masking units 345 and 347 obtain a signal with reduced noise by using the attenuation gain values and the energy ratio values transmitted from the phase difference attenuation gain calculator 340 and the energy ratio calculator 350. By applying the values to Equation 6, a signal with reduced noise can be obtained. The noise-reduced signal is output to the inverse FFT (IFFT) 350 and 352.

The time domain converters 350 and 352 convert the signal in the frequency domain output from the masking units 345 and 347 into a signal in the time domain and then output the signal to the combiner 355.

The combiner 355 removes residual noise by combining the signals which are reduced in noise and converted into the time domain. The combination includes a simple combination of signals through selection of a signal or independent component analysis.

4 is a flowchart illustrating a noise canceling process according to a preferred embodiment of the present invention.

Referring to FIG. 4, in operation 410, two microphones each receive signals.

In step 420, the frequency domain converter converts the signals into signals in the frequency domain.

In step 430, the phase unit and the magnitude unit calculate a phase value and a magnitude value for each frequency from the converted signals.

In step 440, the phase difference calculator and the attenuation gain calculator calculate phase difference and attenuation gain, respectively, using the phase values calculated in step 430. Here, the phase difference may be calculated through Equation 7 by receiving a delay value due to a difference between the mounting positions of the two microphones from the phase compensator. By compensating for the phase difference due to the distance between microphones through the delay value, only the signal estimated as noise remains. The attenuation gain calculator calculates the attenuation gain by inputting the phase difference provided from the phase difference calculator to Equation (3).

In step 450, the energy ratio calculator calculates an energy ratio for each microphone channel using the magnitude value for each frequency calculated in step 430. The energy ratio of each microphone channel means an energy ratio of a signal received through each microphone, and is calculated using Equation 4 or Equation 5. Here, the process of calculating the phase difference and the sensitivity gain in step 440 and the energy ratio for each microphone channel in step 450 are performed in parallel through the devices shown in FIG. 3.

In operation 460, the masking unit removes noise from the converted signal using the attenuation gain values calculated in operations 440 and 450 and the energy ratio of each microphone channel.

In operation 470, the time domain converter converts the signal from which the noise is removed from the frequency domain to a signal in the time domain.

5 is a graph illustrating a signal received through two microphones and a signal from which noise is removed according to a preferred embodiment of the present invention.

Referring to FIG. 5, CH 1 (K) 501 shows a signal input through microphone 1 302 of FIG. 3, and CH 2 (K) 502 is input through microphone 2 304. Show the signal.

Y1 (K) 503 shows a signal from which noise is removed from an input signal received through the microphone 1 302 according to the noise canceling method of the present invention described above. The Y1 (K) 503 signal may be obtained through Equation 6.

Y2 (K) 504 shows a signal from which noise is removed from the input signal received through microphone 2 304 according to the noise cancellation method of the present invention, such as Y1 (K) 503. The Y2 (K) 504 signal may also be obtained through Equation 6.

As shown in the Y1 (K) 503 and Y2 (K) 504 signals of FIG. 5, noise is almost eliminated and only voice remains as compared with signals input to the microphones. Thus, the sound quality for communication can be improved, and the signal-to-noise ratio can be improved by increasing the energy of the signal.

 While the present invention has been described in connection with what is presently considered to be the most practical and preferred embodiment, it is to be understood that the invention is not limited to the disclosed embodiments, but is capable of various modifications within the scope of the invention. Therefore, the scope of the present invention should not be limited by the illustrated embodiments, but should be determined by the scope of the appended claims and equivalents thereof.

In the present invention that operates as described in detail above, the effects obtained by the representative ones of the disclosed inventions will be briefly described as follows.

As described above, the present invention has an advantage that the signal-to-noise ratio estimation is simple and robust to noise removal because the noise level is estimated by the phase information between two signals. Also, because two channels of information can be used regardless of the microphone mounting position, it is possible to overcome the limitation of the conventional two channels of noise cancellation.

Claims (16)

In a portable terminal having a first microphone and a second microphone, the method for removing noise from the signals input to the first microphone and the second microphone, Converting the input signals into signals in a first and a second frequency domain, respectively; Calculating phase values and magnitude values for the signals in the first and second frequency domains; Calculating a difference between the calculated phase values and calculating a gain value in inverse proportion to the difference between the phase values; Obtaining signals in the first and second frequency domains from which noise is removed by multiplying the gain value by the frequency-specific magnitude values of the signals in the first and second frequency domains; And converting the first and second noise canceled signals in the frequency domain into signals in the time domain and outputting the signals. The method of claim 1, wherein the gain value, And multiplying the difference between the phase values by a constant capable of controlling gain and then adding 1 to obtain an inverse. The method of claim 1, before converting the noise canceled first and second frequency domain signals into signals in the time domain. And multiplying energy ratio values for respective microphones by the signals of the first and second frequency domains that have been removed. The method of claim 3, wherein each of the microphone-specific energy ratio values, Calculation is performed by dividing a square value of magnitude values of frequencies for signals in the first or second frequency domain by the sum of squared magnitude values of signals of frequencies in the first and second frequency domains. Noise reduction method characterized in that. The method of claim 1, Selecting one of the signals in the time domain, or separating and combining the signals by an independent component analysis method. The method of claim 1, wherein the difference between the phase values, The phase compensation value for the distance difference between the first microphone and the second microphone is obtained by subtracting the magnitude value of the frequency of the signal of the first frequency domain from the magnitude of the frequency of the signal of the second frequency domain. Noise reduction method characterized in that the value obtained by subtracting the multiplied by the magnitude value for each frequency of the signals of the first and second frequency domain. 7. The method of claim 6, wherein the phase compensation value is a time delay value determined according to a distance difference between the first microphone and the second microphone. The method of claim 1, wherein the first microphone and the second microphone are omnidirectional microphones. In a portable terminal having a first microphone and a second microphone, the apparatus for removing noise from the signals input to the first microphone and the second microphone, A frequency domain converter for converting the input signals into signals of a first and second frequency domain, respectively; A phase unit for calculating phase values for signals in the first and second frequency domains; A magnitude unit for calculating magnitude values of frequencies of the signals of the first and second frequency domains; A phase difference calculator for calculating a difference between the calculated phase values; A gain calculator for calculating a gain value in inverse proportion to the difference between the phase values for the signals in the first and second frequency domains, respectively; A masking unit for multiplying the gain values by the frequency values of the signals in the first and second frequency domains, respectively, to obtain the signals in the first and second frequency domains from which noise is removed; And a time domain converter for converting the first and second noise canceled frequency domain signals into time domain signals and outputting the converted signals. The method of claim 9, wherein the gain value, And calculating the inverse of the difference between the phase values by multiplying a constant capable of controlling gain of the gain and adding 1 to the inverse. 10. The method of claim 9, wherein the masking unit is configured to convert the signals of the first and second frequency domains from which the noise is removed into signals in the time domain before the signals are removed. Noise canceller, characterized in that each of the energy ratio values are further multiplied. The method of claim 11, wherein the energy ratio value for each microphone, It is calculated by dividing the square value of the magnitude value of each frequency with respect to the signal of the first or second frequency domain by the sum of the squared magnitude values of the frequency values of the signals of the first and second frequency domain. Noise canceller characterized in that. 10. The method of claim 9, And a combiner for selecting one of the signals in the time domain or separating and combining the signals in an independent component analysis method. The method of claim 9, wherein the difference between the phase values, The phase compensation value for the distance difference between the first microphone and the second microphone is obtained by subtracting the magnitude value of the frequency of the signal of the first frequency domain from the magnitude of the frequency of the signal of the second frequency domain. Noise canceling device, characterized in that the value obtained by subtracting the multiplied by the magnitude value for each frequency of the signals in the first and second frequency domain. 15. The apparatus of claim 14, wherein the phase compensation value is a predetermined value according to a distance difference between the first microphone and the second microphone, and a time delay value. 10. The apparatus of claim 9, wherein the first microphone and the second microphone are omnidirectional microphones.

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