WO2015184893A1 - Procédé et dispositif de réduction de bruit d'appel vocal pour terminal mobile - Google Patents

Procédé et dispositif de réduction de bruit d'appel vocal pour terminal mobile Download PDF

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Publication number
WO2015184893A1
WO2015184893A1 PCT/CN2015/074770 CN2015074770W WO2015184893A1 WO 2015184893 A1 WO2015184893 A1 WO 2015184893A1 CN 2015074770 W CN2015074770 W CN 2015074770W WO 2015184893 A1 WO2015184893 A1 WO 2015184893A1
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Prior art keywords
microphone
sound source
mobile terminal
microphones
call mode
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PCT/CN2015/074770
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English (en)
Chinese (zh)
Inventor
王进军
孙焘
薛华
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中兴通讯股份有限公司
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Publication of WO2015184893A1 publication Critical patent/WO2015184893A1/fr

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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M1/00—Substation equipment, e.g. for use by subscribers
    • H04M1/72—Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/725—Cordless telephones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones

Definitions

  • This paper relates to the field of mobile communications, and in particular to a method and apparatus for voice noise reduction of a mobile terminal.
  • Voice call is a basic function of mobile terminals.
  • the improvement of voice call quality is a topic that all mobile terminals are committed to.
  • the noise reduction algorithm has evolved from previous single microphone (MIC) noise reduction to dual MIC noise reduction, and even microphone array algorithms.
  • the number of MICs on mobile terminals has also evolved from a single MIC to a now-current dual MIC.
  • Even some mobile terminals have already been 3MIC or 4MIC layouts.
  • the signal-to-noise ratio of the main MIC is required to be 6 dB larger than the signal-to-noise ratio of the sub-MIC.
  • 1 is a schematic diagram of a dual MIC layout manner in the prior art. As shown in FIG. 2, due to the vocal characteristics of a conventional receiver, when the end user uses the handset mode to talk, the main MIC is closer to the mouth than the sub MIC, so It is easy to meet the requirements of the noise reduction algorithm.
  • the terminal user uses the hands-free mode to make a call
  • the signal-to-noise ratio of the primary and secondary MICs changes, so that the signal-to-noise ratio difference between the primary and secondary MICs cannot meet the requirements of the noise reduction algorithm.
  • the noise reduction algorithm is invalidated, and the noise reduction performance of the call is drastically deteriorated. Therefore, a method is needed to identify such changes, and the noise reduction performance of the voice call is ensured, thereby improving the user experience and increasing the market competitiveness of the product.
  • Embodiments of the present invention provide a method and apparatus for voice noise reduction of a mobile terminal.
  • a mobile terminal call voice noise reduction method includes:
  • the microphone that is closer to the sound source is set as the primary microphone, and the other microphone is set as the secondary microphone;
  • the noise reduction algorithm is used to perform noise reduction processing on the call voice of the mobile terminal.
  • the foregoing method further includes:
  • the mobile terminal When the mobile terminal enters the call mode, it is determined by the voice path whether the call mode of the mobile terminal is a hands-free call mode, a hand-held call mode, or a headset call mode, and if it is determined to be a hand-held call mode or a headset call mode, the voice call is directly performed, if When it is determined that the hands-free call mode is activated, the primary and secondary microphone determination operations are started.
  • the calculating the delay of data collected by the two microphones of the mobile terminal includes:
  • the collected data of the two microphone samples are respectively filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of speech spectra respectively corresponding to the two microphones;
  • ⁇ corresponds to the maximum value of R12( ⁇ ) is the time delay ⁇ 12 between two microphones
  • x1(kT) s(kT- ⁇ 1)+n1(kT)
  • x2(kT) s(kT- ⁇ 2 ) +n2(kT)
  • s(kT) is the sound source signal
  • n1(kT) is the background noise of the first microphone
  • n2(kT) is the background noise of the second microphone
  • ⁇ 1 and ⁇ 2 are the sound waves from the sound source to
  • the delay between the data collected, E[] represents the autocorrelation function between x1(kT) and x2(kT).
  • determining, according to the calculated delay, the relative location between the sound source of the mobile terminal user and the two microphones includes:
  • ⁇ is the delay of the data collected by the two microphones
  • c is the speed of sound
  • d is the difference in sound path
  • the elevation angle of the sound source Determine the relative position between the source of the mobile terminal user and the two microphones.
  • the microphone that is closer to the sound source is set as the primary microphone, and the other microphone is set as the secondary microphone, including:
  • a mobile terminal call voice noise reduction device includes:
  • the obtaining module is configured to: obtain a delay of data collected by two microphones of the mobile terminal;
  • Determining a module configured to: determine a relative position between the sound source of the mobile terminal user and the two microphones according to the delay;
  • Set the module set to: set the microphone closer to the sound source as the primary microphone and the other microphone as the secondary microphone according to the relative position;
  • the noise reduction module is configured to: perform noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the primary microphone and the secondary microphone.
  • the foregoing apparatus further includes:
  • the judging module is configured to: when the mobile terminal enters the call mode, determine, by using the voice path, whether the call mode of the mobile terminal is a hands-free call mode, a handheld call mode, or a headset call mode, such as If it is determined to be the handheld call mode or the headset call mode, the voice call is directly performed, and if it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are started.
  • the above obtaining module is set to:
  • the collected data of the two microphone samples are respectively filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of speech spectra respectively corresponding to the two microphones;
  • ⁇ corresponds to the maximum value of R12( ⁇ ) is the time delay ⁇ 12 between two microphones
  • x1(kT) s(kT- ⁇ 1)+n1(kT)
  • x2(kT) s(kT- ⁇ 2 ) +n2(kT)
  • s(kT) is the sound source signal
  • n1(kT) is the background noise of the first microphone
  • n2(kT) is the background noise of the second microphone
  • ⁇ 1 and ⁇ 2 are the sound waves from the sound source to
  • E[] represents the autocorrelation function between x1(kT) and x2(kT).
  • the determining module is configured to:
  • ⁇ is the delay of the data collected by the two microphones
  • c is the speed of sound
  • d is the difference in sound path
  • the elevation angle of the sound source Determine the relative position between the source of the mobile terminal user and the two microphones.
  • the above setting module is set to:
  • a computer readable storage medium storing program instructions that, when executed, implement the methods described above.
  • the primary and secondary MICs of the mobile terminal are dynamically adjusted according to the relative position changes of the user and the mobile terminal, and the noise reduction algorithm is implemented in the related art when the terminal user uses the hands-free mode for talking.
  • the problem of failure can improve the sound quality of hands-free call reception voice without increasing the design cost and space of the mobile communication terminal.
  • the noise reduction performance of the mobile terminal is maintained in a better state, thereby improving the voice quality of the call and improving the user experience.
  • FIG. 1 is a schematic diagram of a dual MIC layout manner in the related art
  • FIG. 2 is a flowchart of a method for reducing voice of a call of a mobile terminal according to an embodiment of the present invention
  • FIG. 3 is a basic basic principle diagram of a far field sound source localization according to an embodiment of the present invention.
  • FIG. 4 is a schematic diagram of calculating a delay of data collected by two microphones according to an embodiment of the present invention
  • FIG. 5 is a flow chart showing the detailed processing of the embodiment of the present invention.
  • FIG. 6 is a flow chart showing the judgment of the primary and secondary MICs according to an embodiment of the present invention.
  • FIG. 7 is a schematic structural diagram of a call voice noise reduction apparatus of a mobile terminal according to an embodiment of the present invention.
  • the invention provides a method and a device for reducing voice of a mobile terminal call voice, which are described below with reference to the accompanying drawings. Embodiments of the present invention will be described in detail. It is understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
  • FIG. 2 is a flowchart of a mobile terminal call voice noise reduction method according to an embodiment of the present invention. As shown in FIG. 2, according to an embodiment of the present invention, The mobile terminal call voice noise reduction method includes the following processing:
  • Step 201 Obtain a delay of data collected by two microphones of the mobile terminal.
  • step 201 when the mobile terminal enters the call mode, it is determined by the voice path whether the call mode of the mobile terminal is a hands-free call mode, a hand-held call mode, or a headset call mode, if it is determined In the handheld call mode or the headset call mode, the voice call is directly performed. If it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are initiated (ie, steps 201-204 are performed).
  • step 201 calculating the delay of data collected by the two microphones of the mobile terminal may include:
  • Step 1 respectively, the collected data of the two microphone samples are filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of voices respectively corresponding to the two microphones.
  • Step 2 Calculate power spectrum data of two sets of audio spectrums, and perform frequency domain weighting, and perform inverse fast Fourier transform IFFT after the power spectrum data is accumulated to a predetermined number of frames, and obtain an autocorrelation function;
  • Step 3 Calculate the delay of the data collected by the two microphones according to Equation 1 based on the calculated autocorrelation function:
  • ⁇ corresponds to the maximum value of R12( ⁇ ) is the time delay ⁇ 12 between two microphones
  • x1(kT) s(kT- ⁇ 1)+n1(kT)
  • x2(kT) s(kT- ⁇ 2 ) +n2(kT)
  • s(kT) is the sound source signal
  • n1(kT) is the background noise of the first microphone
  • n2(kT) is the background noise of the second microphone
  • ⁇ 1 and ⁇ 2 are the sound waves from the sound source to
  • E[] represents the autocorrelation function between x1(kT) and x2(kT).
  • Step 202 Determine a relative position between the sound source of the mobile terminal user and the two microphones according to the delay; and include the following processing:
  • Step 1 Calculate the elevation angle of the sound source when the microphone array is used as the reference coordinate according to Formula 2.
  • ⁇ is the delay of the data collected by the two microphones
  • c is the speed of sound
  • d is the difference in sound path
  • Step 2 according to the elevation angle of the sound source Determine the relative position between the source of the mobile terminal user and the two microphones.
  • Step 203 according to the relative position, the microphone that is closer to the sound source is set as the primary microphone, and the other microphone is set as the secondary microphone;
  • the microphone located at the bottom of the mobile terminal is closer to the sound source when the elevation angle of the sound source It can be determined that the microphone located at the top of the mobile terminal is closer to the sound source;
  • Step 204 Perform noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the primary microphone and the secondary microphone.
  • FIG. 3 is a basic basic principle diagram of a far-field sound source localization according to an embodiment of the present invention.
  • the mobile phone is The relative position of the microphone and the sound source can be regarded as the far field range, so the acoustic signal can be regarded as being transmitted in the form of a plane wave.
  • a horizontal angle can be obtained. That is to say, the delay of the signal received by the two microphones can be used to calculate and determine the orientation of the sound source.
  • the correlation function R12( ⁇ ) of x1(kT) and x2(kT) can be expressed as:
  • the ⁇ corresponding to the maximum value of R12( ⁇ ) is the time delay ⁇ 12 between the two microphones.
  • the elevation angle of the sound source that is, the relative position between the sound source and the microphone, can be obtained from the time delay R12 between the two microphones.
  • the delay between the two microphones of the mobile phone can be calculated by using the existing microphone array and the processing chip of the mobile phone.
  • FIG. 4 is a schematic diagram of calculating a delay of data collected by two microphones according to an embodiment of the present invention.
  • the voice data sampled by the microphone is first filtered by a bandpass filter of 300 to 4 kHz to remove high frequency noise, and then The filtered speech signal is subjected to fast Fourier transform to obtain the speech spectrum; then the power spectrum data (cross-power spectrum) of the signal is obtained for the two sets of audio spectra obtained by the two microphones, and frequency domain weighting is performed, and the power spectrum data is accumulated.
  • the inverse Fourier transform is used to find the autocorrelation function.
  • the obtained autocorrelation function is used to obtain the delay of the data collected by the two microphones.
  • the relative position between the sound source (speaker) and the microphone can be obtained according to the delay of the microphone, and the microphone closer to the sound source (speaker) is set as the main MIC, and the other MIC is set as the sub MIC, thereby Guaranteed noise reduction performance and improved hands-free calling sound quality.
  • FIG. 5 is a flowchart of detailed processing of an embodiment of the present invention. As shown in FIG. 5, the following specifically includes the following processing:
  • Step 501 The mobile terminal enters a call mode.
  • Step 502 The voice path is used to determine whether the mobile terminal is in the hands-free mode or the handheld or headset mode. If the call is in the handheld or headset mode, step 503 is performed to directly perform a voice call. If the voice is in the hands-free mode, step 504 is performed.
  • Step 503 directly performing a voice call
  • Step 504 starting a primary and secondary MIC determination module
  • Step 505 the process shown in FIG. 6 includes the following steps: Step 601, MIC1 and MIC2 receive a voice signal from a sound source (speaker); Step 602, using the above calculation method to calculate when the sound source reaches MIC1 and MIC2 ⁇ 12; Step 603, after obtaining the delay between MIC1 and MIC2, the formula can be utilized Calculate the elevation angle between the sound source and MIC1 and MIC2 Finally, according to the elevation angle The relative position between the sound source (speaker) and the MIC can be obtained when the elevation angle When determining that the MIC1 in Figure 1 is closer to the sound source (speaker), when the elevation angle At this time, the MIC2 distance sound source (speaker) in Fig. 1 is determined; in step 604, the main and sub MICs are determined.
  • the MIC that is closer to the sound source (speaker) is set as the main MIC. If the main MIC setting of the original call is consistent with the calculated main MIC, the execution is performed. In step 507, if the main MIC setting of the original call does not match the calculated main MIC, steps 508 and 509 are performed.
  • Step 507 returning to the voice call
  • Step 508 adjusting a noise reduction algorithm
  • step 509 a voice call is returned.
  • the primary and secondary MIC information used in the hands-free call is constant, so when the relative position of the person and the terminal changes, the signal-to-noise ratio of the primary and secondary MIC may deteriorate, thereby affecting the noise reduction of the mobile terminal. Performance and voice quality of the call.
  • the technical solution of the embodiment of the invention enables the user to update the primary and secondary MIC settings in real time when the position of the person changes with the relative position of the mobile terminal during the hands-free call, thereby keeping the noise reduction algorithm in a stable state and ensuring that the noise reduction algorithm is always in a stable state.
  • the noise reduction performance of the mobile terminal compensates for the influence of the change of the position of the person on the performance of the noise reduction algorithm, thereby improving the sound quality.
  • FIG. 7 is a schematic structural diagram of a mobile terminal call voice noise reduction device according to an embodiment of the present invention.
  • the mobile terminal call voice noise reduction device includes: an obtaining module 70, a determining module 72, a setting module 74, and a noise reduction module 76.
  • the following is a detailed description of each module of the embodiment of the present invention. Detailed instructions.
  • the obtaining module 70 is configured to obtain a time delay of data collected by the two microphones of the mobile terminal; the obtaining module 70 is configured to:
  • the collected data of the two microphone samples are respectively filtered by the band pass filter to remove the high frequency noise, and the filtered collected data is subjected to fast Fourier transform FFT to calculate two sets of speech spectra respectively corresponding to the two microphones;
  • ⁇ corresponds to the maximum value of R12( ⁇ ) is the time delay ⁇ 12 between two microphones
  • x1(kT) s(kT- ⁇ 1)+n1(kT)
  • x2(kT) s(kT- ⁇ 2 ) +n2(kT)
  • s(kT) is the sound source signal
  • n1(kT) is the background noise of the first microphone
  • n2(kT) is the background noise of the second microphone
  • ⁇ 1 and ⁇ 2 are the sound waves from the sound source to
  • E[] represents the autocorrelation function between x1(kT) and x2(kT).
  • the determining module 72 is configured to determine a relative position between the sound source of the mobile terminal user and the two microphones according to the calculated time delay; the determining module 72 may be configured to:
  • ⁇ is the delay of the data collected by the two microphones
  • c is the speed of sound
  • d is the difference in sound path
  • the elevation angle of the sound source Determine the relative position between the source of the mobile terminal user and the two microphones.
  • the setting module 74 is configured to set the microphone closer to the sound source as the primary microphone and the other microphone as the secondary microphone according to the determined relative position; the setting module 74 may be configured as:
  • the noise reduction module 76 is configured to perform noise reduction processing on the call voice of the mobile terminal by using a noise reduction algorithm based on the determined primary microphone and the secondary microphone.
  • the determining module further includes: when the mobile terminal enters the call mode, determining, by using the voice path, whether the call mode of the mobile terminal is a hands-free call mode, a handheld call mode, or a headset call mode, if it is determined to be handheld In the call mode or the headset call mode, the voice call is directly performed, and if it is determined to be the hands-free call mode, the primary and secondary microphone determination operations are started.
  • the primary and secondary MICs of the mobile terminal are dynamically adjusted according to the relative position changes of the user and the mobile terminal, and the noise reduction algorithm is implemented in the related art when the terminal user uses the hands-free mode for talking.
  • the problem of failure can improve the sound quality of hands-free call reception voice without increasing the design cost and space of the mobile communication terminal.
  • the noise reduction performance of the mobile terminal is maintained in a better state, thereby improving the voice quality of the call and improving the user experience.
  • all or part of the steps of the above embodiments may also be implemented by using an integrated circuit. These steps may be separately fabricated into individual integrated circuit modules, or multiple modules or steps may be fabricated into a single integrated circuit module. achieve.
  • the devices/function modules/functional units in the above embodiments may be implemented by a general-purpose computing device, which may be centralized on a single computing device or distributed over a network of multiple computing devices.
  • each device/function module/functional unit in the above embodiment When each device/function module/functional unit in the above embodiment is implemented in the form of a software function module and sold or used as a stand-alone product, it can be stored in a computer readable storage medium.
  • the above mentioned computer readable storage medium may be a read only memory, a magnetic disk or an optical disk or the like.
  • the embodiment of the invention keeps the noise reduction performance of the mobile terminal in a better state, thereby improving the voice quality of the call and improving the user experience.

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Abstract

L'invention concerne un procédé et un dispositif de réduction de bruit d'appel vocal pour terminal mobile, ce procédé comprenant les étapes suivantes consistant à : obtenir des retards d'acquisition de données de deux microphones d'un terminal mobile ; en fonction de ces retards, déterminer les positions relatives entre une source sonore d'un utilisateur de terminal mobile et les deux microphones ; en fonction de ces positions relatives, régler le microphone le plus proche de la source sonore en tant que microphone principal et régler l'autre microphone en tant que microphone auxiliaire ; sur la base du microphone principal et du microphone auxiliaire, effectuer le traitement de réduction du bruit sur un appel vocal du mobile terminal par le biais d'un algorithme de réduction du bruit.
PCT/CN2015/074770 2014-11-21 2015-03-20 Procédé et dispositif de réduction de bruit d'appel vocal pour terminal mobile WO2015184893A1 (fr)

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Cited By (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN105872898A (zh) * 2016-04-12 2016-08-17 北京奇虎科技有限公司 基于双话筒的降噪处理方法及装置
CN107592600A (zh) * 2016-07-06 2018-01-16 深圳市三诺声智联股份有限公司 一种基于分布式麦克风的拾音筛选方法及拾音装置
CN109087648A (zh) * 2018-08-21 2018-12-25 平安科技（深圳）有限公司 柜台语音监控方法、装置、计算机设备及存储介质
WO2022143119A1 (fr) * 2020-12-29 2022-07-07 华为技术有限公司 Procédé de collecte de sons, dispositif électronique et système
CN117768816A (zh) * 2023-11-15 2024-03-26 兴科迪科技（泰州）有限公司 一种基于小尺寸pcba实现声音采集方法及装置

Families Citing this family (10)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
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CN110166879B (zh) 2019-06-28 2020-11-13 歌尔科技有限公司 语音采集控制方法、装置及tws耳机
CN111510903B (zh) * 2020-04-27 2023-06-09 广州市一横电子有限公司 麦克风与移动终端交互方法、存储介质及麦克风智能装置
CN113726947B (zh) * 2020-05-26 2022-09-09 Oppo广东移动通信有限公司 语音通话方法、装置、终端及存储介质
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Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20050239516A1 (en) * 2004-04-27 2005-10-27 Clarity Technologies, Inc. Multi-microphone system for a handheld device
CN101771923A (zh) * 2009-01-07 2010-07-07 王青云 一种眼镜数字助听器声源定位方法
CN102165521A (zh) * 2008-09-30 2011-08-24 苹果公司 多个麦克风切换和配置

Family Cites Families (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN102033223B (zh) * 2010-12-29 2012-10-03 北京信息科技大学 使用麦克风阵列对声源定位的方法
CN103219012B (zh) * 2013-04-23 2015-05-13 中国人民解放军总后勤部军需装备研究所 一种基于声源距离的双麦克风噪声消除方法及装置
CN104270489A (zh) * 2014-09-10 2015-01-07 中兴通讯股份有限公司 一种从多个麦克风中确定主副麦克风的方法和系统

Patent Citations (3)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US20050239516A1 (en) * 2004-04-27 2005-10-27 Clarity Technologies, Inc. Multi-microphone system for a handheld device
CN102165521A (zh) * 2008-09-30 2011-08-24 苹果公司 多个麦克风切换和配置
CN101771923A (zh) * 2009-01-07 2010-07-07 王青云 一种眼镜数字助听器声源定位方法

Cited By (7)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
CN105872898A (zh) * 2016-04-12 2016-08-17 北京奇虎科技有限公司 基于双话筒的降噪处理方法及装置
CN107592600A (zh) * 2016-07-06 2018-01-16 深圳市三诺声智联股份有限公司 一种基于分布式麦克风的拾音筛选方法及拾音装置
CN107592600B (zh) * 2016-07-06 2024-04-02 深圳市三诺声智联股份有限公司 一种基于分布式麦克风的拾音筛选方法及拾音装置
CN109087648A (zh) * 2018-08-21 2018-12-25 平安科技（深圳）有限公司 柜台语音监控方法、装置、计算机设备及存储介质
CN109087648B (zh) * 2018-08-21 2023-10-20 平安科技（深圳）有限公司 柜台语音监控方法、装置、计算机设备及存储介质
WO2022143119A1 (fr) * 2020-12-29 2022-07-07 华为技术有限公司 Procédé de collecte de sons, dispositif électronique et système
CN117768816A (zh) * 2023-11-15 2024-03-26 兴科迪科技（泰州）有限公司 一种基于小尺寸pcba实现声音采集方法及装置

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