<!DOCTYPE article PUBLIC "-//NLM//DTD JATS (Z39.96) Journal Archiving and Interchange DTD v1.0 20120330//EN" "JATS-archivearticle1.dtd">
<article xmlns:xlink="http://www.w3.org/1999/xlink">
  <front>
    <journal-meta />
    <article-meta>
      <title-group>
        <article-title>Differential Microphone Array Speech Enhancement Based on Post - Filtering</article-title>
      </title-group>
      <contrib-group>
        <contrib contrib-type="author">
          <string-name>Quan Trong The</string-name>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Digital Agriculture Cooperative</string-name>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Cau Giay</string-name>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Ha Noi</string-name>
        </contrib>
        <contrib contrib-type="author">
          <string-name>Viet Nam.</string-name>
        </contrib>
      </contrib-group>
      <abstract>
        <p>Noise suppression has become an essential requirement for numerous acoustic devices, mobile phones and surveillance equipment. One standard criterion for almost digital signal processing is the saving of the target speech component while eliminating all background noise and interferences. Dual - microphone system is one of the most basic elements, which is widely commonly installed in speech applications. However, the designed signal processing faces many complex challenges in extracting the directional sound source. In this paper, the author proposed using an additive post-filtering to improve the author's previous research's performance. The evaluated experiment has confirmed the desired noise reduction to 5.5 (dB) and increasing the speech quality in terms of the signal-to-noise ratio from 3.2 (dB) to 5.7 (dB).</p>
      </abstract>
      <kwd-group>
        <kwd>1 microphone array</kwd>
        <kwd>the signal-to-noise ratio</kwd>
        <kwd>speech enhancement</kwd>
        <kwd>noise reduction</kwd>
        <kwd>dual microphone system</kwd>
        <kwd>post - filtering</kwd>
      </kwd-group>
    </article-meta>
  </front>
  <body>
    <sec id="sec-1">
      <title>1. Introduction</title>
      <p>The demand of noise suppression in almost speech processing applications like speech recognition,
hearing aids, cell phone, surveillance equipment, smart home applications to work anytime and
anywhere, makes them has the capacity of reducing the effect of annoying disturbances, like
background noise, interferences, third - party or surrounding vehicle transport. To decrease the
degradation of desired target talker in terms of speech quality, speech intelligibility, the single and multi
- microphone noise reduction approach are often applied for signal processing. Spectral estimation
technique is the most widely implemented in single - channel techniques, such as: subspace method,
Wiener filter and spectral subtraction, are based on calculation of the noise power with assumption that
the background noise are stationary, or under the situation with ambient speech noise.</p>
      <p>
        The microphone array beamforming [
        <xref ref-type="bibr" rid="ref1 ref2 ref3 ref4 ref5 ref6 ref7 ref8 ref9">1-20</xref>
        ] exploits the priori spatial information of the direction of
speaker, the properties of recording environment, the direction of arrival of useful interest signal and
the geometry of MA’s configuration. Due to the possibility to perform spatial beamforming, MA’s
algorithm has a better ability to alleviate the noise level and save the speech component. For
multimicrophone noise suppression, using spatial information to obtain the advantage of preserving the target
signal while eliminating noisy environment. Besides, MA beamforming can be applied the pre
processing, post - filtering methods to achieve noise reduction and decrease the speech distortion.
      </p>
      <p>Dual - microphone system (DMA2) has numerous advantages for signal processing. DMA2 owns
compact, easy to implement MA digital signal processing. In the previous work [23], the author
suggested an effective method for separating each target, which stands at two opposite directions.
However, in the real-life world, due to its undetermined recording situations, the performance also
corrupted. In this contribution, the author proposed using an additive post - filtering to block the
remaining noisy component after using [23]. The numerical results have rated the better performance
of noise suppression to 5.5 (dB) and increasing of the speech quality from 3.2 to 5.7 (dB). The purpose
of this article is demonstrating a new effective technique to enhance speech enhancement, which based
on DMA2.</p>
      <p>The remaining section of this article is organized as follows. The next section describes the signal
model of DMA2 and the author’s previous evaluation. Section III demonstrated additive post - Filtering,
which uses MMSE estimator. Section IV shows the illustrated experiment and Section V concludes the
purpose of this paper.</p>
    </sec>
    <sec id="sec-2">
      <title>2. The signal model</title>
      <p>A scheme of principal working of a certain differential microphone array (DMA2) [24 - 30] is
illustrated in Figure 4. DMA2 owns high directional beampattern, high noise reduction and easy
implemented to form a beampattern toward the sound source. DMA2 has a compact size, and is very
suitable for microphone array technique, digital signal processing method to block surrounding noise
environment and save the target desired speaker.</p>
      <p>With the definition  ,  is the frequency index and current considered frame. We denote the speech
propagation of sound source is  (343  / ),  is distance between two installed microphones,  0 =
 / is the sound delay, the direction of arrival of useful signal is  ,   =   0 ( ). The
representation of two captured microphone array signals in the frequency-domain defined can be
expressed as the following way:
 1( ,  ) =  ( ,  )   
 2( ,  ) =  ( ,  ) −  
(1)
(2)</p>
      <p>With a defined time, delay  is added, the directivity pattern of the processed signal is obtained by
determined value of  . DMA2 is used for extracting two directional different speakers at opposite
directions. The output of DMA, which exploits subtraction signal between  1( ,  ),  2( ,  ) can be
illustrated that:</p>
      <p>The value of   ( )is limited with a determined threshold 12(dB). This equalizer ensures deriving
desired signal.</p>
      <p>So finally, the received signals are:
 1( , )=  1
 2( , )=  2
( , )×   ( )
( , )×   ( )
(10)
(11)
(12)
(13)
(14)</p>
    </sec>
    <sec id="sec-3">
      <title>3. The suggested post - Filtering</title>
      <p>estimator [21] is used for estimation a spectral gain:</p>
      <p>The central ideal of suggested post - Filtering is based on the estimation of noise power. The MMSE
  1( , )=
√ ( , )
 ( , ) [Г(1 +

2</p>
      <p>) (− 2;1;− ( , ))]

1
 ( , )+1
Where  ( , )≜  ( , )</p>
      <p>( , ) ;  ( ; ; )is the confluent hypergeometric function. The author
proposed the calculation of a priori SNR  ( , ) and a posteriori SNR  ( , ) as the following
equations:
 ( , )=</p>
      <p>[| 1( , )|2]
 [| 2</p>
      <p>( , )|2]
 ( , )=  [|[|2 1( , )|2]</p>
      <p>( , )|2]</p>
      <p>With a defined appropriate value  , the obtained gain function by MMSE estimator can be used as
an effective post - Filtering. In single - channel approach, the   1
( , )is applied to gain the desired
speech component while suppressing noise level. In the next section, this post - Filtering has the ability
of preserving the target speaker while decreasing the background noise and enhancing the overall
performance.</p>
    </sec>
    <sec id="sec-4">
      <title>4. Experiments</title>
      <p>In this section, the author illustrated an experiment to enhance the performance of DMA2 in noise
reduction. This experiment is conducted in a living room, where in presence of annoying background
noise, diffuse noise field. The speaker in stand at  = 2( ) to the DMA2. For further to rate the
performance, an objective measurement [22] is used for calculating the speech quality of the previous
work and additive post - Filtering. Two microphone array signals are sampled at  = 16  , and
transformed in the frequency domain with these parameters:  = 512, overlap 50%.</p>
      <p>The waveform of the original microphone array signal can be expressed in Figure 7. From 0 - 1.4
(s), the speech component of desired talker exits, and from 1.6 - 3 (s), there are only direction noise
source.</p>
      <p>Using [23], the obtained waveform is shown in Figure 8.</p>
      <p>By using the post - Filtering, the effectiveness of noise reduction can be obtained. The processed
signal is shown in Figure 9.</p>
      <p>The overall energy of microphone array signal, the processed signals by [23], and post – Filtering is
depicted in Figure 10.</p>
      <p>As we can see that, the advantage of post - Filtering is presented. The achieved noise reduction is to
5.5 (dB), and the speech quality in the terms of the signal - to - noise ratio (SNR) is increased from 3.2
(dB) to 5.7 (dB).</p>
      <p>An essential problem is almost signal processing is achievement of more robust noise reduction.
DMA2 is one of the most widely common installed in numerous speech applications, such as mobile
phone, surveillance equipment, smart home, teleconferencing due to its compact. Therefore, an
effective post - Filtering for DMA2 is an attractive research direction. In this section, the improvement
of noise reduction has been confirmed. The numerical results show that proposed post - Filtering allows
obtaining better performance in DMA2.</p>
    </sec>
    <sec id="sec-5">
      <title>5. Conclusion</title>
      <p>DMA2 has the capacity of compact arrangements, low size and promise high directional
beampattern, high gain the output signal in comparison with other MA beamforming. However, DMA
owns its drawback that even the complex noisy environment can corrupt its performance. For many
several speech applications, decreasing speech distortion or noise suppression is always a considered
problem. In this research, the author has presented and demonstrated a post - Filtering for enhancing
the DMA2’s performance in realistic recording scenario. The author has shown how the noisy
component at certain direction can be suppressed, and the speech quality of DMA2’s evaluation was
increased in comparison with the author’s previous work. This post - Filtering can be integrated into
multi-microphone system, which use other different MA beamforming.</p>
    </sec>
    <sec id="sec-6">
      <title>6. Acknowledgements</title>
      <p>This research was supported by Digital Agriculture Cooperative. The author thanks our colleagues
from Digital Agriculture Cooperative, who provided insight and expertise that greatly assisted the
research.</p>
    </sec>
    <sec id="sec-7">
      <title>7. References</title>
      <p>[10] Wang X., Cohen I., Benesty J., Chen J. Study of the Null Directions on The Performance
of Differential Beamformers. ICASSP 2022 - 2022 IEEE International Conference on Acoustics,
Speech and Signal Processing (ICASSP). DOI: 10.1109/ICASSP43922.2022.9746462.
[11] Tu Q., Chen H. Theoretical Lower Bounds on the Performance of the
FirstOrder Differential Microphone Arrays With Sensor Imperfections. IEEE/ACM Transactions on
Audio, Speech, and Language Processing. Pp(s): 785 – 801. DOI: 10.1109/TASLP.2022.3145317.
[12] Shi C., Du F., Liu C., Li H. Differential Error Feedback Active Noise Control With the Auxiliary
Filter Based Mapping Method. IEEE Signal Processing Letters. Pp(s): 573 – 577.</p>
      <p>DOI: 10.1109/LSP.2022.3144839.
[13] Itzhak G., Cohen I., Benesty J. Robust Differential Beamforming with Rectangular Arrays // Proc
2021 29th European Signal Processing Conference (EUSIPCO).</p>
      <p>DOI: 10.23919/EUSIPCO54536.2021.9616085.
[14] Zhang P., Gao J., Bian Y., Huang Y. A Preliminary on the Sound Pickup for Unmanned Aerial
Vehicles Using Differential Microphone Arrays. 2021 4th International Conference on
Information Communication and Signal Processing (ICICSP).</p>
      <p>DOI: 10.1109/ICICSP54369.2021.9611979.
[15] Chen Z., Chen H., Tu Q. Sensor Imperfection Tolerance Analysis of Robust
Linear Differential Microphone Arrays. IEEE/ACM Transactions on Audio, Speech, and
Language Processing. Pp(s): 2915 – 2929. DOI: 10.1109/TASLP.2021.3110136.
[16] Yu G., Qiu Y., Wang N. A Robust Wavenumber-Domain Superdirective Beamforming for
Endfire Arrays. IEEE Transactions on Signal Processing Page(s): 4890 – 4905.</p>
      <p>DOI: 10.1109/TSP.2021.3105754.
[17] Huang G., Wang Y., Benesty J., Cohen I., Chen J. Combined Differential Beamforming With
Uniform Linear Microphone Arrays. ICASSP 2021 - 2021 IEEE International Conference on
Acoustics, Speech and Signal Processing (ICASSP). DOI: 10.1109/ICASSP39728.2021.9414189.
[18] Zhao X., Huang G., Benesty J., Chen J., Cohen I. On the Design of
Square Differential Microphone Arrays with a Multistage Structure. ICASSP 2021 - 2021 IEEE
International Conference on Acoustics, Speech and Signal Processing (ICASSP).</p>
      <p>DOI: 10.1109/ICASSP39728.2021.9413759.
[19] Borra F., Bernardini A., Bertuletti I., Antonacci F., Sarti A. Arrays of First-Order
Steerable Differential Microphones. ICASSP 2021 - 2021 IEEE International Conference on
Acoustics, Speech and Signal Processing (ICASSP). DOI: 10.1109/ICASSP39728.2021.9413476.
[20] Wang X., Huang G., Cohen I., Benesty J., Chen J. Robust Steerable Differential Beamformers with
Null Constraints for Concentric Circular Microphone Arrays. ICASSP 2021 - 2021 IEEE
International Conference on Acoustics, Speech and Signal Processing (ICASSP).</p>
      <p>DOI: 10.1109/ICASSP39728.2021.9414119.
[21] Ephraim Y., Malah D. Speech enhancement using minimum mean-square error log-spectral
amplitude estimator. IEEE Trans. ASSP, vol. ASSP-33, no. 2, pp. 443–445, 1985.</p>
      <p>DOI: 10.1109/TASSP.1985.1164550.
[22] https://labrosa.ee.columbia.edu/projects/snreval/
[23] Stolbov M., Tatarnikova M., The Q.T. (2018) Using Dual-Element Microphone Arrays for
Automatic Keyword Recognition // Karpov A., Jokisch O., Potapova R. (eds) Speech and
Computer. SPECOM 2018. Lecture Notes in Computer Science, vol 11096. Springer, Cham.
https://doi.org/10.1007/978-3-319-99579-3-68.
[24] Ma D., Wang Y., He L., Jin M., Su D., Yu D. DP-DWA: Dual-Path Dynamic Weight Attention
Network With Streaming Dfsmn-San For Automatic Speech Recognition. ICASSP 2022 - 2022
IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP).</p>
      <p>DOI: 10.1109/ICASSP43922.2022.9746328.
[25] Xu X., Gu R., Zou Y. Improving Dual-Microphone Speech Enhancement by Learning
CrossChannel Features with Multi-Head Attention. ICASSP 2022 - 2022 IEEE International Conference
on Acoustics, Speech and Signal Processing (ICASSP).</p>
      <p>DOI: 10.1109/ICASSP43922.2022.9746359.
[26] Xiao Z., Chen T., Liu Y., Li J., Li Z. Keystroke Recognition with the Tapping Sound Recorded by
Mobile Phone Microphones. IEEE Transactions on Mobile Computing.</p>
      <p>DOI: 10.1109/TMC.2021.3137229.
[27] Tan K., Zhang X., Wang D.L. Deep Learning Based Real-Time Speech Enhancement for
DualMicrophone Mobile Phones. IEEE/ACM Transactions on Audio, Speech, and Language
Processing. DOI: 10.1109/TASLP.2021.3082318.
[28] Tan K., Zhang X., Wang D.L. Real-Time Speech Enhancement for Mobile Communication Based
on Dual-Channel Complex Spectral Mapping. ICASSP 2021 - 2021 IEEE International
Conference on Acoustics, Speech and Signal Processing (ICASSP).</p>
      <p>DOI: 10.1109/ICASSP39728.2021.9414346.
[29] Shankar N., Bhat G.S., Panahi M.S. Real-time dual-channel speech enhancement by VAD assisted
MVDR beamformer for hearing aid applications using smartphone. 2020 42nd Annual
International Conference of the IEEE Engineering in Medicine &amp; Biology Society (EMBC).</p>
      <p>DOI: 10.1109/EMBC44109.2020.9175212.
[30] Li H., Zhang X., Gao G. Beamformed Feature for Learning-based Dual-channel Speech
Separation. ICASSP 2020 - 2020 IEEE International Conference on Acoustics, Speech and Signal
Processing (ICASSP). DOI: 10.1109/ICASSP40776.2020.9054049.</p>
    </sec>
  </body>
  <back>
    <ref-list>
      <ref id="ref1">
        <mixed-citation>
          [1]
          <string-name>
            <surname>Albertini</surname>
            <given-names>D.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Bernardini</surname>
            <given-names>A.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Borra</surname>
            <given-names>F.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Antonacci</surname>
            <given-names>F.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Sarti</surname>
            <given-names>A</given-names>
          </string-name>
          .
          <article-title>Two-Stage Beamforming With Arbitrary Planar Arrays of Differential Microphone Array Units</article-title>
          . IEEE/ACM Transactions on Audio, Speech,
          <string-name>
            <given-names>and Language</given-names>
            <surname>Processing</surname>
          </string-name>
          . Pp:
          <fpage>590</fpage>
          -
          <lpage>602</lpage>
          . DOI:
          <volume>10</volume>
          .1109/TASLP.
          <year>2022</year>
          .
          <volume>3231719</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref2">
        <mixed-citation>
          [2]
          <string-name>
            <surname>Huang</surname>
            <given-names>W.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Feng J. Robust Steerable</surname>
          </string-name>
          <article-title>Differential Beamformer for Concentric Circular Array With Directional Microphones // Proc 2022 Asia-Pacific Signal and Information Processing Association Annual Summit and Conference (APSIPA ASC)</article-title>
          .
          <source>DOI: 10.23919/APSIPAASC55919</source>
          .
          <year>2022</year>
          .
          <volume>9980184</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref3">
        <mixed-citation>
          [3]
          <string-name>
            <given-names>JWang J.</given-names>
            ,
            <surname>Yang</surname>
          </string-name>
          <string-name>
            <given-names>F.</given-names>
            ,
            <surname>Yang</surname>
          </string-name>
          <string-name>
            <surname>J</surname>
          </string-name>
          .
          <article-title>Insights Into the MMSE-Based Frequency-Invariant Beamformers for Uniform Circular Arrays</article-title>
          .
          <source>IEEE Signal Processing Letters</source>
          . Pp(s):
          <fpage>2432</fpage>
          -
          <lpage>2436</lpage>
          . DOI:
          <volume>10</volume>
          .1109/LSP.
          <year>2022</year>
          .
          <volume>3224687</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref4">
        <mixed-citation>
          [4]
          <string-name>
            <surname>Ueno</surname>
            <given-names>N.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Kameoka</surname>
            <given-names>H</given-names>
          </string-name>
          .
          <source>Multiple Sound Source Localization Based on Stochastic Modeling of Spatial Gradient Spectral // Proc 2022 30th European Signal Processing Conference (EUSIPCO)</source>
          .
          <source>DOI: 10.23919/EUSIPCO55093</source>
          .
          <year>2022</year>
          .
          <volume>9909524</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref5">
        <mixed-citation>
          [5]
          <string-name>
            <surname>Yan</surname>
            <given-names>L.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Huang</surname>
            <given-names>W.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Kleijn</surname>
            <given-names>W.B.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Abhayapala</surname>
            <given-names>T. D.</given-names>
          </string-name>
          <article-title>Phase Error Analysis for First-Order Linear Differential Microphone Arrays /</article-title>
          / Proc 2022 International Workshop on Acoustic Signal Enhancement (IWAENC).
          <source>DOI: 10.1109/IWAENC53105</source>
          .
          <year>2022</year>
          .
          <volume>9914748</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref6">
        <mixed-citation>
          [6]
          <string-name>
            <surname>Itzhak</surname>
            <given-names>G.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Cohen</surname>
            <given-names>I.</given-names>
          </string-name>
          <article-title>Differential and Constant-Beamwidth Beamforming with Uniform Rectangular Arrays</article-title>
          .
          <source>2022 International Workshop on Acoustic Signal Enhancement (IWAENC)</source>
          .
          <source>DOI: 10.1109/IWAENC53105</source>
          .
          <year>2022</year>
          .
          <volume>9914769</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref7">
        <mixed-citation>
          [7]
          <string-name>
            <surname>Huang</surname>
            <given-names>G.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Benesty</surname>
            <given-names>J.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Chen</surname>
            <given-names>J</given-names>
          </string-name>
          .
          <article-title>Fundamental Approaches to Robust Differential Beamforming With High Directivity Factors</article-title>
          . IEEE/ACM Transactions on Audio, Speech,
          <string-name>
            <given-names>and Language</given-names>
            <surname>Processing</surname>
          </string-name>
          . Pp(s):
          <fpage>3074</fpage>
          -
          <lpage>3088</lpage>
          . DOI:
          <volume>10</volume>
          .1109/TASLP.
          <year>2022</year>
          .
          <volume>3209935</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref8">
        <mixed-citation>
          [8]
          <string-name>
            <surname>Jin</surname>
            <given-names>J.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Benesty</surname>
            <given-names>J.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Huang</surname>
            <given-names>H.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Chen J. On</surname>
          </string-name>
          <article-title>Differential Beamforming With Nonuniform Linear Microphone Arrays</article-title>
          . IEEE/ACM Transactions on Audio, Speech,
          <string-name>
            <given-names>and Language</given-names>
            <surname>Processing</surname>
          </string-name>
          . Pp(s):
          <fpage>1840</fpage>
          -
          <lpage>1852</lpage>
          . DOI:
          <volume>10</volume>
          .1109/TASLP.
          <year>2022</year>
          .
          <volume>3178229</volume>
          .
        </mixed-citation>
      </ref>
      <ref id="ref9">
        <mixed-citation>
          [9]
          <string-name>
            <surname>Yang</surname>
            <given-names>X.</given-names>
          </string-name>
          ,
          <string-name>
            <surname>Wei</surname>
            <given-names>J. DMANET</given-names>
          </string-name>
          :
          <article-title>Deep Learning-Based Differential Microphone Arrays for MultiChannel Speech Separation</article-title>
          .
          <source>ICASSP 2022 - 2022 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)</source>
          .
          <source>DOI: 10.1109/ICASSP43922</source>
          .
          <year>2022</year>
          .
          <volume>9747725</volume>
          .
        </mixed-citation>
      </ref>
    </ref-list>
  </back>
</article>