WO2014101429A1 - 一种终端双麦克风降噪的方法及装置 - Google Patents

一种终端双麦克风降噪的方法及装置 Download PDF

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Publication number
WO2014101429A1
WO2014101429A1 PCT/CN2013/081335 CN2013081335W WO2014101429A1 WO 2014101429 A1 WO2014101429 A1 WO 2014101429A1 CN 2013081335 W CN2013081335 W CN 2013081335W WO 2014101429 A1 WO2014101429 A1 WO 2014101429A1
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Prior art keywords
noise reduction
terminal
actual distance
microphone
distance
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PCT/CN2013/081335
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English (en)
French (fr)
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邢蓓蕾
胡蝶
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中兴通讯股份有限公司
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Publication of WO2014101429A1 publication Critical patent/WO2014101429A1/zh

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    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • G10L21/0216Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02165Two microphones, one receiving mainly the noise signal and the other one mainly the speech signal

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a method and a related device for noise reduction of a terminal dual microphone.
  • Dual microphone wind noise reduction technology refers to the built-in two microphones in the mobile phone, which realizes the noise suppression function by collecting the voice signal and environmental noise of the call, as shown in Figure 1.
  • the main microphone simultaneously receives the voice signal of the call and the background noise signal; and the secondary microphone mainly collects the noise in the environment, and after processing the signal waveform, is used to synthesize the sample signal of the main microphone, so that the receiver of the call is made. I only hear the call voice, but I can't hear the ambient noise, and realize the dual microphone noise reduction function.
  • the secondary microphone in the dual microphone collects background noise, including static noise, dynamic noise, and cell phone echo noise. Based on a large amount of experimental data in various complex environments, the mobile phone manufacturer sets parameters for the noise processing module in the mobile phone, which can effectively achieve the dual microphone noise reduction effect on the mobile phone and improve the communication quality.
  • the traditional two-microphone noise reduction method usually uses fixed parameters in the noise reduction module. This allows the mobile phone to achieve excellent noise reduction under fixed conditions of use, as shown in Figure 2 and Figure 3.
  • the distance between the main microphone and the source usually the person's mouth
  • the noise reduction function loses its effect.
  • the distance between the primary microphone and the sound source is greater than or equal to the distance between the secondary microphone and the sound source, normal voice signals will be suppressed by the noise reduction module, affecting normal voice calls, as shown in Figure 4 and Figure 5. Show.
  • the embodiment of the invention provides a method and a device for reducing noise of a dual microphone of a terminal, which can adjust the parameters of the dual microphone noise reduction processing module in real time by dynamically collecting the handheld gesture information, thereby improving the call quality.
  • a method for reducing noise of a terminal dual microphone includes: During the terminal call, the handheld gesture information of the terminal is collected in real time;
  • the call audio is subjected to noise reduction processing according to the adaptive noise reduction parameter.
  • the initial noise reduction parameter is a secondary microphone initial gain
  • the adaptive noise reduction parameter is a secondary microphone adaptive gain
  • Auxmicgain k ⁇ Aumicgain 0
  • Auxmiclen 0 - Mainmiclen 0 wherein the Main c/e «, Auxmiclen, Auxmicgain are respectively the first actual ⁇ giant away, the second actual ⁇ giant away, the secondary microphone adaptive gain, M? nmiclen 0 , Auxmiclen 0 ⁇ Auxmicgain ⁇ is divided into a first standard distance, a second standard distance, and a secondary microphone initial gain, and k is an adjustment coefficient.
  • the secondary microphone adaptive gain is adjusted to zero.
  • the handheld gesture information collection module is configured to: during the terminal call, the handheld gesture information of the terminal is collected in real time;
  • a distance calculation module configured to: obtain, by using the handheld posture information, a first actual distance between the terminal main microphone and the sound source, a second actual distance between the terminal secondary microphone and the sound source; and a dual microphone noise reduction processing module, The method is configured to: perform real-time adjustment on the initial noise reduction parameter in the terminal by using the first actual distance and the second actual distance, obtain an adaptive noise reduction parameter, and perform a call according to the adaptive noise reduction parameter.
  • the audio is noise-reduced.
  • the dual microphone noise reduction processing module comprises: Comparing the submodule, the setting is: comparing the first actual distance and the second actual distance;
  • a noise reduction parameter calculation submodule configured to: when the first actual distance is smaller than the second real
  • Microphone adaptive gain wherein the Main c/ew, Auxmiclen, Auxmicgain are 1 J for the first actual large separation, the second actual large separation, the secondary microphone adaptive gain, Mammiclen 0 , Auxmiclen 0 uxmicgain 0 respectively A standard distance, a second standard distance, a secondary microphone fixed gain, k is an adjustment coefficient, the initial noise reduction parameter is a secondary microphone initial gain, and the adaptive noise reduction parameter is a secondary microphone adaptive gain.
  • the noise reduction parameter calculation sub-module is further configured to: when the first actual distance is greater than or equal to the second actual distance, adjust the secondary microphone adaptive gain to zero.
  • the embodiment of the invention overcomes the limitation of the traditional fixed noise reduction parameter dual microphone noise reduction technology, dynamically adjusts the initial noise reduction parameter by collecting the hand posture information of the terminal during the call, and realizes the dual microphone noise reduction optimization technology, thereby Eliminating the disadvantages of traditional noise reduction schemes in some cases, and even affecting normal voice quality.
  • 1 is a schematic diagram of a terminal dual microphone
  • Figure 2 is a front elevational view of a standard handheld gesture handset in a call state
  • Figure 3 is a side view of a standard handheld gesture handset in a call state
  • Figure 4 is a schematic diagram of the actual hand-held posture in the call state (front and rear rotation);
  • Figure 5 is a schematic diagram of the actual hand-held posture in the call state (outward rotation);
  • FIG. 6 is a flowchart of a method for reducing noise of a terminal dual microphone according to an embodiment of the present invention
  • FIG. 7 is a block diagram of a device for reducing noise of a dual microphone of a terminal according to an embodiment of the present invention. Preferred embodiment of the invention
  • FIG. 6 is a flowchart of a method for reducing noise of a dual microphone of a terminal according to an embodiment of the present invention. As shown in FIG. 6, the method includes:
  • Step 601 During the terminal call, the handheld gesture information of the terminal is collected in real time.
  • Step 602 Acquire, according to the handheld posture information, a first actual distance between the terminal main microphone and the sound source, and a second actual distance between the terminal secondary microphone and the sound source.
  • Step 603 Perform real-time adjustment on the initial noise reduction parameter in the terminal by using the first actual distance and the second actual distance to obtain an adaptive noise reduction parameter.
  • the initial noise reduction parameter is a secondary microphone initial gain
  • the adaptive noise reduction parameter is a secondary microphone adaptive gain. Comparing the first actual distance and the second actual distance, and adjusting the secondary microphone adaptive gain to zero when the first actual distance is greater than or equal to the second actual distance, when the first When the actual distance is less than the second actual distance, the secondary microphone adaptive gain is calculated by the following formula:
  • Auxmicgain k ⁇ Aumicgain 0
  • Auxmiclen 0 - Mainmiclen 0 wherein the Main c/e «, Auxmiclen, Auxmicgain are respectively the first actual ⁇ giant away, the second actual ⁇ giant away, the secondary microphone adaptive gain, M? nmiclen 0 , Auxmiclen 0 ⁇ Auxmicgain ⁇ is divided into a first standard distance, a second standard distance, a secondary microphone initial gain, and k is an adjustment coefficient.
  • Step 604 Perform noise reduction processing on the call audio according to the adaptive noise reduction parameter.
  • FIG. 7 is a block diagram of a device for reducing noise of a dual microphone according to an embodiment of the present invention. As shown in FIG. 7, the method includes: a handheld gesture information collection module, a distance calculation module, and a dual microphone noise reduction processing module, wherein:
  • the handheld gesture information collection module is configured to: collect handheld gesture information of the terminal in real time during the terminal call, the handheld gesture information includes an actual three-dimensional angle of the terminal, and the gyroscope and the gravity sensor of the module are collected by using the handheld posture information
  • the device collects the actual three-dimensional angle, that is, the handheld gesture information collection module collects the actual handheld state of the terminal during the terminal call (eg, The state shown in Fig. 4 or Fig. 5 is an angle change with respect to the standard hand posture (the state shown in Figs. 2 and 3).
  • the distance calculation module is configured to: obtain, by using the handheld posture information, a first actual distance between the terminal main microphone and the sound source, and a second actual distance between the terminal secondary microphone and the sound source.
  • the dual microphone noise reduction processing module is configured to: perform real-time adjustment on the initial noise reduction parameter in the terminal by using the first actual distance and the second actual distance to obtain an adaptive noise reduction parameter, and according to the adaptation Noise reduction parameters, noise reduction processing of call audio.
  • the initial noise reduction parameter is a secondary microphone initial gain
  • the adaptive noise reduction parameter is a secondary microphone adaptive gain.
  • the comparison sub-module of the dual-microphone noise reduction processing module compares the first actual distance obtained by the adaptive noise reduction parameter calculation module with the second actual distance, so as to adjust the initial microphone initial gain in real time, and obtain the secondary microphone self Adapt to the gain.
  • the noise reduction parameter calculation submodule of the dual microphone noise reduction processing module passes the formula when the first actual distance is smaller than the second actual distance
  • Auxmicgain Auxrmclen - Mamrmclen . k .
  • Aumicgain 0 calculates the secondary microphone adaptive gain
  • Step 1 Figure 2 and Figure 3 are the front view of the mobile phone in the standard hand-held posture and the side view of the mobile phone in the call state, respectively, assuming that the terminal earpiece is close to the ear, the secondary microphone is near the earpiece, and the main microphone is close to the mouth under the standard hand posture.
  • the terminal length is L.
  • the distance between the main microphone and the sound source is the first standard distance MaiiTM C / e «.
  • the distance between the secondary microphone and the sound source is the second standard distance Auxmiclen, and the standard three-dimensional angle of the terminal is c, and the terminal uses the secondary microphone initial gain Auxmicgain ⁇ to perform noise reduction processing.
  • Step 2 When the handheld posture of the terminal changes, as shown in FIG. 4 or FIG. 5, the actual state of the terminal during the call is recorded through a device such as a gyroscope and a gravity sensor built in the terminal.
  • the three-dimensional angle 2 , ⁇ 2 is used to determine the first actual distance Mainmiclen of the main microphone and the sound source of the terminal in the three-dimensional space, namely:
  • the end of the terminal main microphone rotates back and forth (as shown in Fig. 4) or outward (as shown in Fig. 5), so the second actual distance of the secondary microphone and the sound source can be approximated by Auxmic!
  • the second standard distance is Auxmic!en 0 , which is the terminal length L.
  • Step 3 according to the previous step Dynamically adjust the secondary microphone adaptive gain Auxmicgain to maintain the call clarity of the terminal.
  • the Main c/e «, Auxmiclen, Auxmicgain are respectively ll the first actual ⁇ macro, the second actual ⁇ macro, the secondary microphone adaptive gain; M? nmiclen 0 , Auxmiclen 0 ⁇ Auxmicgain ⁇ separate
  • the first standard distance, the second standard distance, and the secondary microphone initial gain; k is an adjustment coefficient, which is determined by the test result, and the adjustment coefficients of different items are different; the second standard distance is approximately equal to the terminal length L.
  • Embodiment 2 is a diagrammatic representation of Embodiment 1:
  • Step 1 Figure 2 and Figure 3 are the front view of the mobile phone in the standard handheld posture and the side view of the mobile phone in the call state, respectively, assuming that the terminal earpiece is close to the ear, the vertical distance between the secondary microphone and the earpiece is h, under the standard hand posture, the main The microphone is close to the mouth, the terminal length is L, and the vertical distance between the handset and the main microphone is L.
  • the distance between the main microphone and the sound source is the first standard distance Mainmiclen
  • the distance between the secondary microphone and the sound source is the second standard distance Auxmic!en 0
  • the standard three-dimensional angle of the terminal is, A,
  • the terminal uses the secondary microphone initial Gain Awoni Cg ain. Perform noise reduction processing.
  • a shorter side of the terminal is taken as the X axis, and a longer side is taken as the Y axis to form a plane rectangular coordinate system.
  • the main microphone, the secondary microphone, and the earpiece are arranged.
  • the coordinates are ( ⁇ ⁇ ), ( ⁇ 2 , ⁇ 2 ). ( ⁇ 3 , ⁇ 3 ).
  • the vertical distance h between the secondary microphone and the earpiece is -ysl, and the vertical distance between the earpiece and the main microphone. It is
  • the actual three-dimensional angle of the terminal during the call is recorded by using a device such as a gyroscope and a gravity sensor built in the terminal, 2 , ⁇ , Thereby determining the first actual distance Ma ww c/e" and the second actual distance ⁇ a /cfe" of the main microphone and the sound source of the terminal in the three-dimensional space, namely:
  • Step 3 according to the previous step Dynamically adjust the secondary microphone adaptive gain Auxmicgain to maintain the call clarity of the terminal.
  • the Main c/e «, Auxmiclen, Auxmicgain are respectively ll the first actual ⁇ macro, the second actual ⁇ macro, the secondary microphone adaptive gain; M? nmiclen 0 , Auxmiclen 0 ⁇ Auxmicgain ⁇ separate
  • the first standard distance, the second standard distance, and the secondary microphone initial gain; k is an adjustment coefficient, which is determined by the test result, and the adjustment coefficients of different items are different; the second standard distance is approximately equal to the terminal length L.
  • Step 4. Repeat steps 2 and 3 until the end of the call.
  • the embodiment of the present invention can adjust the initial noise reduction parameter in the terminal in real time according to the position and angle of the handheld gesture, thereby reducing the ambient noise and effectively improving the speech clarity of the terminal. Improve call quality.
  • modules or steps of the present invention can be implemented by a general-purpose computing device, which can be concentrated on a single computing device or distributed over a network composed of multiple computing devices. Alternatively, they may be implemented by program code executable by the computing device so that they may be stored in the storage device by the computing device, or they may be separately fabricated into individual integrated circuit modules, or Multiple modules or steps are made into a single integrated circuit module. Thus, the invention is not limited to any specific combination of hardware and software. Although the invention has been described in detail above, the invention is not limited thereto, and various modifications may be made by those skilled in the art in accordance with the principles of the invention. Therefore, modifications made in accordance with the principles of the invention should be construed as falling within the scope of the invention.
  • the initial noise reduction parameter in the terminal is adjusted in real time, so as to reduce the ambient noise, the speech clarity of the terminal can be effectively improved, and the call quality is improved.

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Abstract

一种终端双麦克风降噪的方法及装置,涉及通信技术领域,所述方法包括:终端通话期间,实时采集终端的手持姿态信息;根据所述手持姿态信息,获取终端主麦克风与声源间的第一实际距离、终端次麦克风与声源间的第二实际距离;利用所述第一实际距离和所述第二实际距离,对终端中的初始降噪参数进行实时调整,得到自适应降噪参数;根据所述自适应降噪参数,对通话音频进行降噪处理。

Description

一种终端双麦克风降噪的方法及装置 技术领域
本发明涉及通信技术领域, 特别涉及一种终端双麦克风降噪的方法及相 关装置。
背景技术
手机等通讯设备上釆用双麦克风降噪技术, 已经成为主流趋势。 双麦克 风降噪技术, 是指手机中内置两个麦克风, 通过釆集通话的语音信号和环境 噪声, 实现噪声抑制的功能, 如图 1 所示。 其中, 主麦克风同时接收通话的 语音信号和背景噪声信号; 而次麦克风则主要收集环境中的噪声, 对信号波 形进行处理后, 用来与主麦克风的釆样信号进行合成, 让通话的接收方只听 到通话语音, 而听不到环境噪声, 实现双麦克风降噪功能。
双麦克风中的次麦克风釆集背景噪声, 包括静态噪声、 动态噪声和手机 回响噪声。 手机制造商依据各种复杂环境下的大量实验数据, 对手机中的噪 声处理模块进行参数设置, 可以在手机上有效的实现双麦克风降噪效果, 提 高通信质量。
然而, 传统的双麦克风降噪方法, 通常在降噪模块使用固定参数。 这使 得手机在固定的使用条件下可以达到优异的降噪效果, 如图 2和图 3所示。 一旦在通话过程中调整手机的手持状态, 导致主麦克风和声源 (通常是人的嘴 巴)的距离不断变化, 降噪功能即会失去效果。 某些情况下, 如果主麦克风和 声源的距离大于等于次麦克风和声源的距离, 就会导致正常的语音信号也会 被降噪模块抑制, 影响正常语音通话, 如图 4和图 5所示。
发明内容
本发明实施例提供一种终端双麦克风降噪的方法及装置, 通过动态地釆 集手持姿态信息, 实时调整双麦克风降噪处理模块的参数, 提高通话质量。
本发明实施例提供的一种终端双麦克风降噪的方法, 包括: 终端通话期间, 实时釆集终端的手持姿态信息;
根据所述手持姿态信息, 获取终端主麦克风与声源间的第一实际距离、 终端次麦克风与声源间的第二实际距离;
利用所述第一实际距离和所述第二实际距离, 实时对终端中的初始降噪 参数进行实时调整, 得到自适应降噪参数;
根据所述自适应降噪参数, 对通话音频进行降噪处理。
优选地, 所述初始降噪参数是次麦克风初始增益, 所述自适应降噪参数 是次麦克风自适应增益。
优选地, 比较所述第一实际距离和所述第二实际距离的大小, 并当所述 第一实际距离小于所述第二实际距离时, 通过以下公式计算所述次麦克风自 适应增益:
. . . Auxmiclen-Mainmiclen , , . .
Auxmicgain = k · Aumicgain0
Auxmiclen0 -Mainmiclen0 其中 , 所述 Main c/e«, Auxmiclen, Auxmicgain分另 ll为第一实际 ^巨离、 第二 实际^巨离、次麦克风自适应增益 , M? nmiclen0 , Auxmiclen0 ^Auxmicgain^分另 ll为第一 标准距离、 第二标准距离和次麦克风初始增益, k为调节系数。
优选地, 当所述第一实际距离大于等于所述第二实际距离时, 将所述次 麦克风自适应增益调整为零。
本发明实施例提供的一种终端双麦克风降噪的装置, 包括:
手持姿态信息釆集模块, 其设置为: 终端通话期间, 实时釆集终端的手 持姿态信息;
距离计算模块, 其设置为: 利用所述手持姿态信息, 获取终端主麦克风 与声源间的第一实际距离、 终端次麦克风与声源间的第二实际距离; 以及 双麦克风降噪处理模块, 其设置为: 利用所述第一实际距离和所述第二 实际距离, 对终端中的初始降噪参数进行实时调整, 得到自适应降噪参数, 并根据所述自适应降噪参数, 对通话音频进行降噪处理。
优选地, 所述双麦克风降噪处理模块包括: 比较子模块, 其设置为: 比较所述第一实际距离和所述第二实际距离; 以及
降噪参数计算子模块, 其设置为: 当所述第一实际距离小于所述第二实
Figure imgf000004_0001
麦克风自适应增益, 其中, 所述 Main c/ew, Auxmiclen, Auxmicgain分另1 J为第一实 际巨离、第二实际巨离、次麦克风自适应增益, Mammiclen0 , Auxmiclen0 uxmicgain0 分别为第一标准距离、 第二标准距离、 次麦克风固定增益, k为调节系数, 所 述初始降噪参数是次麦克风初始增益, 所述自适应降噪参数是次麦克风自适 应增益。 优选地, 所述降噪参数计算子模块还设置为: 当所述第一实际距离大于 等于所述第二实际距离时, 将所述次麦克风自适应增益调整为零。
本发明实施例克服了传统的固定降噪参数双麦克风降噪技术的局限性, 通过釆集终端在通话过程中的手持姿态信息, 动态调整初始降噪参数, 实现 双麦克风降噪优化技术,从而消除了传统的降噪方案在某些情况下效果下降, 甚至影响正常语音质量的缺点。 附图概述
图 1是终端双麦克风示意图;
图 2是通话状态下 标准手持姿态手机的正面示意图;
图 3是通话状态下 标准手持姿态手机的侧面示意图;
图 4是通话状态下 实际手持姿态示意图 (前后转动) ;
图 5是通话状态下 实际手持姿态示意图 (向外转动) ;
图 6是本发明实施例提供的终端双麦克风降噪的方法流程图;
图 7是本发明实施例提供的终端双麦克风降噪的装置框图。 本发明的较佳实施方式
以下结合附图对本发明的优选实施例进行详细说明, 需要说明的是, 在 不冲突的情况下, 本申请中的实施例及实施例中的特征可以相互任意组合。
图 6是本发明实施例提供的终端双麦克风降噪的方法流程图, 如图 6所 示, 该方法包括:
步骤 601、 终端通话期间, 实时釆集终端的手持姿态信息。
步骤 602、 根据所述手持姿态信息, 获取终端主麦克风与声源间的第一 实际距离、 终端次麦克风与声源间的第二实际距离。
步骤 603、 利用所述第一实际距离和所述第二实际距离, 对终端中的初 始降噪参数进行实时调整, 得到自适应降噪参数。
所述步骤 603中, 所述初始降噪参数是次麦克风初始增益, 所述自适应 降噪参数是次麦克风自适应增益。 比较所述第一实际距离和所述第二实际距 离, 并当所述第一实际距离大于等于所述第二实际距离时, 将所述次麦克风 自适应增益调整为零, 当所述第一实际距离小于所述第二实际距离时, 通过 以下公式计算所述次麦克风自适应增益:
. . . Auxmiclen-Mainmiclen , , . .
Auxmicgain = k · Aumicgain0
Auxmiclen0 -Mainmiclen0 其中 , 所述 Main c/e«, Auxmiclen, Auxmicgain分另 ll为第一实际^巨离、 第 二实际 ^巨离、次麦克风自适应增益 , M? nmiclen0 , Auxmiclen0 ^Auxmicgain^分另 ll为第 一标准距离、 第二标准距离、 次麦克风初始增益, k为调节系数。
步骤 604、 根据所述自适应降噪参数, 对通话音频进行降噪处理。
图 7是本发明实施例提供的终端双麦克风降噪的装置框图,如图 7所示, 包括: 手持姿态信息釆集模块、 距离计算模块和双麦克风降噪处理模块, 其 中:
所述手持姿态信息釆集模块, 设置为: 终端通话期间, 实时釆集终端的 手持姿态信息, 所述手持姿态信息包括终端的实际三维角度, 利用手持姿态 信息釆集模块的陀螺仪和重力传感器等设备釆集所述实际三维角度, 也就是 说, 所述手持姿态信息釆集模块在终端通话时, 釆集终端实际手持状态 (如 图 4或图 5所示状态)相对于标准手持姿态 (如图 2和图 3所示状态) 的角 度变化。
所述距离计算模块, 设置为: 利用所述手持姿态信息, 获取终端主麦克 风与声源间的第一实际距离、 终端次麦克风与声源间的第二实际距离。
双麦克风降噪处理模块, 设置为: 利用所述第一实际距离和所述第二实 际距离, 对终端中的初始降噪参数进行实时调整, 得到自适应降噪参数, 并 根据所述自适应降噪参数, 对通话音频进行降噪处理。 其中, 所述初始降噪 参数是次麦克风初始增益, 所述自适应降噪参数是次麦克风自适应增益。 所 述双麦克风降噪处理模块的比较子模块将所述自适应降噪参数计算模块得到 的第一实际距离和第二实际距离进行大小比较, 以便实时地调整次麦克风初 始增益, 得到次麦克风自适应增益。 所述双麦克风降噪处理模块的降噪参数 计算子模块在所述第一实际距离小于所述第二实际距离时, 通过公式
Auxmicgain = Auxrmclen-Mamrmclen . k . Aumicgain0计算所述次麦克风自适应增益,
Auxmiclen0 -Mainmiclen0 并在所述第一实际距离大于等于所述第二实际距离时, 将所述次麦克风自适 应增益调整为零, 其中, 所述 Main c/e«, Auxmiclen, Auxmicgain分别为第一实际 巨离、 第二实际巨离、 次麦克风自适应增益 , M? nmiclen0 , Auxmiclen0 uxmicgain0 分别为第一标准距离、 第二标准距离、 次麦克风固定增益, k为调节系数。 实施例一:
步骤 1 , 图 2和图 3分别是通话状态下, 标准手持姿态的手机的正面示 意图和手机的侧面示意图, 假设终端听筒贴近耳朵, 次麦克风在听筒附近, 标准手持姿态下, 主麦克风靠近嘴巴, 终端长度为 L, 此时, 主麦克风与声 源的距离为第一标准距离 Maii™C/e«。,次麦克风与声源的距离为第二标准距离 Auxmiclen, , 终端的标准三维角度为 c , 终端使用次麦克风初始增益 Auxmicgain^进行降噪处理。
步骤 2 , 当终端的手持姿态发生变化时, 如图 4或图 5所示手持状态, 通过终端内置的陀螺仪和重力传感器等设备, 记录终端在通话过程中的实际 三维角度 2, , ^2 , 以此确定三维空间中终端的主麦克风和声源的第一实际距 离 Mainmiclen , 即:
Mainmiclen = ] (L · cos αλ -L · cos<¾ )2 + (L · cos^, -L · cosy52 )2 + (L · cos γλ -L · cos γ2 )2
终端的手持姿态发生变化时, 一般是终端主麦克风所在的一端前后转动 (如图 4 )或向外转动(如图 5 ) , 因此可以将次麦克风与声源的第二实际距 离 Auxmic!en近似为第二标准距离 Auxmic!en0 , 即终端长度 L。
步骤 3 , 依据上一步骤得到的
Figure imgf000007_0001
, 动态调整次麦克风 自适应增益 Auxmicgain , 以保持终端的通话清晰度。
Auxmiclen-Mainmiclen
■ k · Aumicgain0 Mainmiclen < Auxmiclen
Auxmicgain Auxmiclen0 -Mainmiclen^
0 Mainmiclen > Auxmiclen
其中 , 所述 Main c/e«, Auxmiclen, Auxmicgain分另 ll为第一实际^巨离、 第二实 际^巨离、次麦克风自适应增益; M? nmiclen0 , Auxmiclen0 ^Auxmicgain^分另 ll为第一标 准距离、第二标准距离、次麦克风初始增益; k为调节系数, 由测试结果确定, 不同项目的调节系数不同; 所述第二标准距离约等于终端长度 L。 步骤 4、 不断重复步骤 2和步骤 3 , 直到终端通话结束。
实施例二:
步骤 1 , 图 2和图 3分别是通话状态下, 标准手持姿态的手机的正面示 意图和手机的侧面示意图, 假设终端听筒贴近耳朵, 次麦克风与听筒的垂直 距离为 h, 标准手持姿态下, 主麦克风靠近嘴巴, 终端长度为 L, 听筒与主麦 克风的垂直距离为 L。 此时, 主麦克风与声源的距离为第一标准距离 Mainmiclen, , 次麦克风与声源的距离为第二标准距离 Auxmic!en0 , 终端的标准 三维角度为 , A, , 终端使用次麦克风初始增益 AwoniCgain。进行降噪处理。
其中,将终端的一个较短的侧边作为 X轴,将一个较长的侧边作为 Y轴, 形成平面直角坐标系, 在所述坐标系中, 4叚设主麦克风、 次麦克风、 听筒的 坐标分别为(χι )、 (χ22) . (χ33) , 此时, 所述次麦克风与听筒的垂直距离 h 为 -ysl , 所述听筒与主麦克风的垂直距离为 |y3 -」, 可以近似为终端长度 L。 步骤 2, 当终端的手持姿态发生变化时, 如图 4或图 5所示手持状态, 通过终端内置的陀螺仪和重力传感器等设备, 记录终端在通话过程中的实际 三维角度 , 2,^ , 以此确定三维空间中终端的主麦克风和声源的第一实际距 离 Ma ww c/e"和第二实际距离^ a /cfe" , 即:
Mainmiclen = (L · cos αλ -L · cos<¾ )2 + (L · cos^, -L · cosy52 )2 + (L · cos γλ -L · cos γ2 )2 Auxmiclen =
Figure imgf000008_0001
步骤 3、 依据上一步骤得到的
Figure imgf000008_0002
, 动态调整次麦克风 自适应增益 Auxmicgain , 以保持终端的通话清晰度。
Auxmiclen-Mainmiclen
■ k · Aumicgain0 Mainmiclen < Auxmiclen
Auxmicgain Auxmiclen0 -Mainmiclen^
0 Mainmiclen > Auxmiclen
其中 , 所述 Main c/e«, Auxmiclen, Auxmicgain分另 ll为第一实际^巨离、 第二实 际^巨离、次麦克风自适应增益; M? nmiclen0 , Auxmiclen0 ^Auxmicgain^分另 ll为第一标 准距离、第二标准距离、次麦克风初始增益; k为调节系数, 由测试结果确定, 不同项目的调节系数不同; 所述第二标准距离约等于终端长度 L。
步骤 4, 不断重复步骤 2和步骤 3 , 直到终端通话结束。
综上所述, 本发明实施例能够根据终端手持姿态的位置、 角度等信息, 对终端中的初始降噪参数进行实时调整, 实现降低环境噪声的同时, 能够有 效地改善终端的语音清晰度,提高通话质量。
显然, 本领域的技术人员应该明白, 上述的本发明的各模块或各步骤可 以用通用的计算装置来实现, 它们可以集中在单个的计算装置上, 或者分布 在多个计算装置所组成的网络上, 可选地, 它们可以用计算装置可执行的程 序代码来实现, 从而可以将它们存储在存储装置中由计算装置来执行, 或者 将它们分别制作成各个集成电路模块, 或者将它们中的多个模块或步骤制作 成单个集成电路模块来实现。 这样, 本发明不限制于任何特定的硬件和软件 结合。 尽管上文对本发明进行了详细说明, 但是本发明不限于此, 本技术领域 技术人员可以根据本发明的原理进行各种修改。 因此, 凡按照本发明原理所 作的修改, 都应当理解为落入本发明的保护范围。
工业实用性
本发明实施例对终端中的初始降噪参数进行实时调整, 实现降低环境噪 声的同时, 能够有效地改善终端的语音清晰度,提高通话质量。

Claims

权 利 要 求 书
1、 一种终端双麦克风降噪的方法, 包括:
终端通话期间, 实时釆集终端的手持姿态信息;
根据所述手持姿态信息, 获取终端主麦克风与声源间的第一实际距离以 及终端次麦克风与声源间的第二实际距离;
利用所述第一实际距离和所述第二实际距离 , 对终端中的初始降噪参数 进行实时调整, 得到自适应降噪参数; 根据所述自适应降噪参数, 对通话音频进行降噪处理。
2、 根据权利要求 1所述的方法, 其中, 所述初始降噪参数是次麦克风初 始增益, 所述自适应降噪参数是次麦克风自适应增益。
3、 根据权利要求 2所述的方法, 其中, 所述利用所述第一实际距离和所 述第二实际距离, 对终端中的初始降噪参数进行实时调整, 得到自适应降噪 参数的步骤, 包括: 比较所述第一实际距离和所述第二实际距离的大小, 并 当所述第一实际距离小于所述第二实际距离时, 通过以下公式计算所述次麦 克风自适应增益:
. . . Auxmiclen-Mainmiclen , , . .
Auxmicgain = k · Aumicgain0
Auxmiclen0 -Mainmiclen0 其中 , 所述 Main c/e«, Auxmiclen, Auxmicgain分另 ll为第一实际^巨离、 第 二实际 ^巨离、次麦克风自适应增益 , M? nmiclen0 , Auxmiclen0 ^Auxmicgain^分另 ll为第 一标准距离、 第二标准距离和次麦克风初始增益, k为调节系数。
4、 根据权利要求 3所述的方法, 其中, 所述利用所述第一实际距离和所 述第二实际距离, 对终端中的初始降噪参数进行实时调整, 得到自适应降噪 参数的步骤, 还包括: 当所述第一实际距离大于等于所述第二实际距离时, 将所述次麦克风自适应增益调整为零。
5、 一种终端双麦克风降噪的装置, 包括:
手持姿态信息釆集模块, 其设置为: 终端通话期间, 实时釆集终端的手 持姿态信息; 距离计算模块, 其设置为: 利用所述手持姿态信息, 获取终端主麦克风 与声源间的第一实际距离、 终端次麦克风与声源间的第二实际距离; 以及 双麦克风降噪处理模块, 其设置为: 利用所述第一实际距离和所述第二 实际距离, 对终端中的初始降噪参数进行实时调整, 得到自适应降噪参数, 并根据所述自适应降噪参数, 对通话音频进行降噪处理。
6、根据权利要求 5所述的装置,其中,所述双麦克风降噪处理模块包括: 比较子模块, 其设置为: 比较所述第一实际距离和所述第二实际距离的 大小;
降噪参数计算子模块, 其设置为: 当所述第一实际距离小于所述第二实 际距离时,通过公式 AUXmiCgain = Auxmrclen-Marnmrclen ^謹^謹。计算所述次
A xmiclen0 -Mainmiclen0 麦克风自适应增益, 其中, 所述 Main cfew, Auxmiclen, Auxmicgai 分另1 J为第一实 际 巨 离 、 第 二 实 际 ^巨 离 和 次 麦 克 风 自 适 应 增 益 ,
Msi miclen0 , A xmiclen0 ^ xmicgain0分另 ll为第一标准 ^巨离、第二标准 ^巨离和次麦克 风固定增益, k为调节系数, 所述初始降噪参数是次麦克风初始增益, 所述自 适应降噪参数是次麦克风自适应增益。
7、 根据权利要求 6所述的装置, 其中, 所述降噪参数计算子模块还设置 为: 当所述第一实际距离大于等于所述第二实际距离时, 将所述次麦克风自 适应增益调整为零。
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