WO2015085693A1 - 一种根据周围噪音大小自动切换情景模式的方法及系统 - Google Patents

一种根据周围噪音大小自动切换情景模式的方法及系统 Download PDF

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Publication number
WO2015085693A1
WO2015085693A1 PCT/CN2014/075888 CN2014075888W WO2015085693A1 WO 2015085693 A1 WO2015085693 A1 WO 2015085693A1 CN 2014075888 W CN2014075888 W CN 2014075888W WO 2015085693 A1 WO2015085693 A1 WO 2015085693A1
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background noise
data
microphone
scene mode
buzzer
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French (fr)
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陈琼
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Huizhou TCL Mobile Communication Co Ltd
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Huizhou TCL Mobile Communication Co Ltd
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/72Mobile telephones; Cordless telephones, i.e. devices for establishing wireless links to base stations without route selection
    • H04M1/724User interfaces specially adapted for cordless or mobile telephones
    • H04M1/72448User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions
    • H04M1/72454User interfaces specially adapted for cordless or mobile telephones with means for adapting the functionality of the device according to specific conditions according to context-related or environment-related conditions

Definitions

  • the present invention relates to the field of volume control technologies, and in particular, to a method and system for automatically switching a scene mode according to a surrounding noise level.
  • an object of the present invention is to provide a method and system for automatically switching a scene mode according to the surrounding noise level, aiming at solving the problem that the current mobile terminal volume and scene mode cannot automatically switch with ambient noise and ambient noise. Determine the problem of inaccuracy.
  • a method for automatically switching a scenario according to a surrounding noise level includes:
  • the scene mode is switched according to the complete background noise data.
  • the method for automatically switching a scene mode according to a surrounding noise level wherein the external sound signal data includes background noise signal data and user call sound data, the first microphone is mainly used to acquire background noise signal data, and the second The microphone is mainly used to obtain the voice data of the user's call.
  • the method for automatically switching a scene mode according to a surrounding noise level wherein the first microphone, the second microphone, and the buzzer are respectively disposed at different positions of the mobile terminal, the first microphone and the
  • the second microphone is a microelectromechanical system microphone.
  • the step of switching the scene mode according to the complete background noise data is specifically: matching the complete background noise data with a preset scene mode switching list, and applying the complete background noise data to match the corresponding scene mode, wherein In the scene mode switching list, the background noise decibel value is in one-to-one correspondence with the scene mode.
  • the method for automatically switching a scene mode according to a surrounding noise level wherein a first background noise threshold is set in the scene mode switching list, and when the complete background noise data is higher than the first background noise threshold, In the scene mode corresponding to the background noise data, the larger the background noise decibel value, the larger the call volume and the ring volume.
  • the method for automatically switching a scene mode according to a surrounding noise level wherein a second background noise threshold is set in the scene mode switching list, when the complete background noise data is lower than the second background noise threshold, In the scene mode corresponding to the complete background noise data, the smaller the background noise decibel value, the smaller the call volume and the ring volume.
  • the first microphone and the second microphone are used to acquire external sound signal data
  • An audio digital signal coprocessor for analyzing and processing the external sound signal data acquired by the first microphone and the second microphone and the low frequency signal data acquired by the buzzer to obtain complete Background noise data
  • a baseband application processor configured to switch a scenario according to the complete background noise data obtained by the audio digital signal coprocessor
  • the first microphone, the second microphone and the buzzer are connected to the audio digital signal coprocessor, and the audio digital signal coprocessor is connected to the baseband application processor;
  • the audio digital signal coprocessor includes the following modules interconnected in sequence:
  • a preliminary processing data calculation module configured to compare the external sound signal data acquired by the first microphone and the second microphone to eliminate preliminary high-frequency background noise to obtain preliminary processing data
  • a user voice data calculation module configured to compare the preliminary processing data obtained by the preliminary processing data calculation module with the low frequency signal data obtained by the buzzer, and eliminate low frequency noise to obtain user voice data
  • the complete background noise data calculation module is configured to obtain complete background noise data according to the user call voice data obtained by the user call voice data calculation module.
  • the first microphone, the second microphone, and the buzzer are coupled to the audio digital signal coprocessor, and the audio digital signal coprocessor is coupled to the baseband application processor.
  • the system automatically switches the scene mode system according to the surrounding noise level, wherein the system further includes an audio system module for providing an audio digital and analog interface, and the external sound acquired by the first microphone and the second microphone
  • the signal data and the low frequency signal data obtained by the buzzer are converted into digital signals by an analog signal, and the audio system module is respectively connected to the first microphone, the second microphone, and the buzzer, and The audio digital signal coprocessor is connected.
  • the invention provides a method and a system for automatically switching a scene mode according to the surrounding noise level, which realizes the purpose of intercepting pure background noise by setting a technical means of contrast between background noise and call sound, and finally realizes that the mobile terminal does not need manual adjustment.
  • the effect of automatically switching the scene mode according to the background noise ensures that the user has better call quality regardless of the environment.
  • FIG. 1 is a flow chart of a method for automatically switching a scene mode according to a surrounding noise level according to the present invention
  • FIG. 2 is a schematic block diagram of a system for automatically switching a scene mode according to the surrounding noise level according to the present invention
  • FIG. 3 is a schematic block diagram of a system of an audio digital signal coprocessor according to the present invention.
  • FIG. 4 is a schematic diagram showing the arrangement positions of the first and second microphones and the buzzer in the preferred embodiment of the present invention.
  • the present invention provides a method and system for automatically switching a scene mode according to the size of the surrounding noise.
  • the present invention will be further described in detail below in order to make the objects, technical solutions and effects of the present invention more clear and clear. It is understood that the specific embodiments described herein are merely illustrative of the invention and are not intended to limit the invention.
  • a method for automatically switching a scenario according to a surrounding noise level as shown in FIG. 1 wherein the method comprises the following steps:
  • the external sound signal data includes background noise signal data and user voice data
  • the first microphone is mainly used to acquire background noise signal data
  • the second microphone is mainly used to acquire user voice data.
  • the first microphone 110, the second microphone 120, and the buzzer 130 are respectively disposed at different positions of the mobile terminal 100.
  • the second microphone 120 is disposed at a position where the mobile terminal is close to the user's voice, and can acquire most of the user's call voice data and a small portion of the external environment noise signal.
  • the first microphone 110 is disposed at a position where the mobile terminal emits sound away from the user's call. Therefore, the first microphone 110 acquires most of the background noise signal data, and a small portion of the user calls the sound data.
  • the buzzer 130 serves as a sensor, and through its high sensitivity to low frequency signals, assists in acquiring low frequency signal noise data in background noise, such as wind, echo, and the like in an external environment, and the sound signal data acquired by the three includes All sound signals in the peripheral environment make their data as close as possible to the real external ambient noise.
  • the first microphone 110 and the second microphone 120 are micro-electromechanical system microphones (the micro-electromechanical system microphone is a microphone manufactured based on micro-electromechanical system technology, in short, a capacitor is integrated on the micro-silicon wafer, and Fabricated using a surface mount process, it can withstand high reflow temperatures, is easily integrated with complementary metal oxide semiconductor processes and other audio circuits, and has improved noise cancellation performance with good noise and electromagnetic radiation suppression.
  • the micro-electromechanical system microphone is a microphone manufactured based on micro-electromechanical system technology, in short, a capacitor is integrated on the micro-silicon wafer, and Fabricated using a surface mount process, it can withstand high reflow temperatures, is easily integrated with complementary metal oxide semiconductor processes and other audio circuits, and has improved noise cancellation performance with good noise and electromagnetic radiation suppression.
  • the processing process is performed by a preset processing algorithm for the acquired external sound signal data and the low frequency signal data, and the processing implemented by the algorithm separates the background noise from the user's call voice to the maximum extent.
  • the background noise and the human voice are processed more accurately and better by simply analyzing the loudness of the mixed sound of the human voice and the background noise.
  • the loudness threshold is used to distinguish between vocal and background noise. If the vocals are emotional when talking, the loudness of the vocals may exceed the set loudness threshold, resulting in misjudgment of the system.
  • the present invention does not have the above problems.
  • the technical solution of the present invention intercepts pure background noise according to the set algorithm, and further utilizes background noise to further switch the scene mode.
  • the step S300 is specifically: matching the complete background noise data in the step S200 with a preset scene mode switching list, and applying a scene mode corresponding to the matching of the complete background noise data, wherein the scene mode switching In the list, the background noise decibel value corresponds to the scene mode one-to-one.
  • the first background noise threshold is set in the scene mode switching list, and when the full background noise data is higher than the first background noise threshold, in the scene mode corresponding to the complete background noise data, The larger the background noise decibel value, the larger the call volume and the ringtone volume.
  • a second background noise threshold is set in the scene mode switching list, and when the complete background noise data is lower than the second background noise threshold, in the scenario corresponding to the complete background noise data, The smaller the background noise decibel value, the smaller the call volume and the ringer volume.
  • the first background noise threshold is 60 decibels (dB).
  • the first background noise threshold is set to 60. Decibel, if the obtained complete background noise data is higher than the first background noise threshold, the mobile terminal enters a scene mode corresponding to the complete background noise data, in which the call volume and the ring volume are correspondingly increased. In order to ensure that the user can talk properly, and to ensure that the user can hear the ringtone, do not miss the phone.
  • the volume of the call and the ringtone between them may be the same, but if the complete background noise data is too small, it is required Setting a second background noise threshold, when the full background noise data is lower than the second background noise threshold, the volume of the call and the ringtone in the corresponding scene mode are reduced, and the lower the background noise data is, the corresponding The voice volume and ringtone of the scene mode are smaller, which can automatically reduce the voice of the call when the user is in a very quiet environment, ensure that the call content is not leaked, and because the environment is quiet, the ring tone is small. Also enough to remind users.
  • the user can also set the first background noise threshold and the second background noise threshold according to the sensitivity of his own hearing, and set it to be appropriate for the user. Value.
  • the scene mode includes a call volume level and a ringtone volume.
  • different profiles also include settings for other properties, such as vibration switches, signal indicator switches, and so on.
  • the system has a system for automatically switching a scene mode according to the surrounding noise level in the mobile terminal 100, wherein the system includes: a first microphone 110, a second microphone 120, and a buzzer 130.
  • the audio digital signal coprocessor 200 and the baseband application processor 300 are included in the system.
  • the first microphone 110 and the second microphone 120 are configured to acquire external sound signal data, and the buzzer 130 is configured to acquire low frequency signal data.
  • the audio digital signal coprocessor 200 is configured to analyze and process the external sound signal data acquired by the first microphone 110 and the second microphone 120 and the low frequency signal data acquired by the buzzer 130. Obtain complete background noise data;
  • the baseband application processor 300 is configured to switch a scenario according to the complete background noise data obtained by the audio digital signal coprocessor 200;
  • the first microphone 110, the second microphone 120 and the buzzer 130 are connected to the audio digital signal coprocessor 200, and the audio digital signal coprocessor 200 is connected to the baseband application processor 300;
  • the audio digital signal coprocessor 200 includes the following modules that are sequentially connected:
  • the preliminary processing data calculation module 210 the user call voice data calculation module 220, and the complete background noise data calculation module 230.
  • the preliminary processing data calculation module 210 is configured to compare the external sound signal data acquired by the first microphone 110 and the second microphone 120 respectively, and eliminate the medium-high frequency background noise to obtain preliminary processing data.
  • the user voice data calculation module 220 is configured to compare the preliminary processing data acquired by the preliminary processing data calculation module 210 with the low frequency signal data acquired by the buzzer 130 to eliminate low frequency noise. Get user call voice data.
  • the complete background noise data calculation module 230 is configured to obtain complete background noise data according to the user call voice data acquired by the user call voice data calculation module 220.
  • the system further includes an audio system module 400 for providing an audio digital and analog interface, the external sound signal data acquired by the first microphone 110, the second microphone 120, and the buzzer 130 acquired
  • the low frequency signal data is converted from an analog signal to a digital signal.
  • the first microphone 110, the second microphone 120, and the buzzer 130 are connected to the audio digital signal coprocessor 200 through the audio system module 400, and the audio digital signal coprocessor 200 is connected to the The baseband application processor 300 is described.
  • the mobile terminal is provided with a baseband processor 600, a radio frequency transceiver 710, and a radio frequency amplifier 720.
  • the baseband processor 600 is respectively connected to the audio digital signal coprocessor 200 and the audio system module 400.
  • the radio frequency transceiver 710 is connected to the baseband processor 600, the radio frequency amplifier 720 and the radio frequency transceiver.
  • the 710 connection in combination with the system for automatically switching the scene mode according to the surrounding noise level, forms a complete mobile terminal system, so that the mobile terminal system not only has the function of automatically switching the scene mode according to the surrounding noise level, but also has the extracted Pure user call voice data is used in the call process, which is equivalent to setting noise reduction processing for the user's call, which improves the call quality.
  • the invention provides a method and a system for automatically switching a scene mode according to the surrounding noise level, which realizes the purpose of intercepting pure background noise by setting a technical means of contrast between background noise and call sound, and finally realizes that the mobile terminal does not need manual adjustment, and can be
  • the background noise automatically switches the effect of the scene mode to ensure that the user has better call quality regardless of the environment.

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  • Engineering & Computer Science (AREA)
  • Environmental & Geological Engineering (AREA)
  • Human Computer Interaction (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Signal Processing (AREA)
  • Telephone Function (AREA)

Abstract

 本发明通过获取外部声音信号数据及低频信号数据;对所述外部声音信号数据处理消除中高频背景噪音得到初步处理数据,对所述初步处理数据和所述低频信号处理消除低频噪音得到用户通话声音数据,根据用户通话声音数据获得完全背景噪音数据;根据所得到的完全背景噪音数据切换情景模式。可根据背景噪音自动切换情景模式的效果。

Description

一种根据周围噪音大小自动切换情景模式的方法及系统 技术领域
本发明涉及音量控制技术领域,尤其涉及一种根据周围噪音大小自动切换情景模式的方法及系统。
背景技术
随着手持移动设备的快速普及, 以及用户对于终端设备智能化要求进一步加强,怎样使手机更加智能化,并且更加贴近真实生活的使用习惯,已经成为了我们当务之急所要解决的问题。在不同的噪声环境下,用户为了不漏掉一个重要的电话或者短信,较多情况下都通过手动去调节手机自带的情景模式,比如在外部嘈杂的环境下,将情景模式手动调成户外模式,在开会的时候,将情景模式手动调试成会议模式等等。这样操作起来比较繁琐,也不智能。目前也已经出现了消除外部环境噪音提高通话质量的技术方案,但现有的该类技术方案对人声与外部环境噪音的区分并不十分准确,导致了移动终端对通话环境的错误判断,给用户带来一定的困扰。
因此,如何解决目前移动终端音量及情景模式不能自动随环境噪音进行切换以及对环境噪音判断不准确的问题是音量控制技术领域的研究方向之一。
技术问题
鉴于上述现有技术的不足,本发明的目的在于提供一种根据周围噪音大小自动切换情景模式的方法及系统,旨在解决目前移动终端音量及情景模式不能自动随环境噪音进行切换以及对环境噪音判断不准确的问题。
技术解决方案
一种根据周围噪音大小自动切换情景模式的方法,其中,所述方法包括:
通过第一麦克风、第二麦克风分别获取外部声音信号数据,通过蜂鸣器获取低频信号数据;
将所述第一麦克风和第二麦克风获取的所述外部声音信号数据进行对比,消除中高频背景噪音得到初步处理数据;
将所述初步处理数据与所述蜂鸣器获取的所述低频信号数据进行对比,消除低频噪音得到用户通话声音数据;
根据所述用户通话声音数据获得完全背景噪音数据;
根据所述完全背景噪音数据切换情景模式。
所述根据周围噪音大小自动切换情景模式的方法,其中,所述外部声音信号数据包括背景噪音信号数据和用户通话声音数据,所述第一麦克风主要用于获取背景噪音信号数据,所述第二麦克风主要用于获取用户通话声音数据。
所述的根据周围噪音大小自动切换情景模式的方法,其中,所述第一麦克风、所述第二麦克风和所述蜂鸣器分别设置在移动终端的不同位置,所述第一麦克风和所述第二麦克风为微型机电系统麦克风。
所述根据周围噪音大小自动切换情景模式的方法,其中,
根据所述完全背景噪音数据切换情景模式的步骤具体为:将所述完全背景噪音数据与预先设置的情景模式切换列表进行匹配,应用所述完全背景噪音数据所匹配对应的情景模式,其中,所述情景模式切换列表中,所述背景噪声分贝值与所述情景模式一一对应。所述根据周围噪音大小自动切换情景模式的方法,其中,所述情景模式中包括通话音量大小和铃声音量大小。
所述根据周围噪音大小自动切换情景模式的方法,其中,所述情景模式切换列表中设置第一背景噪音阈值,当所述完全背景噪音数据高于所述第一背景噪音阈值时,与该完全背景噪音数据所对应的所述情景模式中,所述背景噪音分贝值越大,所述通话音量和所述铃声音量越大。
所述根据周围噪音大小自动切换情景模式的方法,其中,所述第一背景噪音阈值为60 分贝。
所述根据周围噪音大小自动切换情景模式的方法,其中,所述情景模式切换列表中设置第二背景噪音阈值,当所述完全背景噪音数据低于所述第二背景噪音阈值的时,与该完全背景噪音数据所对应的所述情景模式中,所述背景噪音分贝值越小,所述通话音量和所述铃声音量越小。
一种具有如上所述的根据周围噪音大小自动切换情景模式系统,其中,所述系统包括:
第一麦克风、第二麦克风用于获取外部声音信号数据;
蜂鸣器,用于获取低频信号数据;
音频数字信号协处理器,用于对所述第一麦克风、所述第二麦克风所获取的所述外部声音信号数据及所述蜂鸣器所获取的所述低频信号数据进行分析处理,获得完全背景噪音数据;
基带应用处理器,用于根据所述音频数字信号协处理器所得到的所述完全背景噪音数据切换情景模式;
所述第一麦克风、所述第二麦克风及所述蜂鸣器与所述音频数字信号协处理器连接,所述音频数字信号协处理器连接所述基带应用处理器;
所述音频数字信号协处理器其包括以下依次互连接的模块:
初步处理数据计算模块,用于将所述第一麦克风和第二麦克风所获取的所述外部声音信号数据进行对比,消除中高频背景噪音得到初步处理数据;
用户通话声音数据计算模块,用于将所述初步处理数据计算模块得到的所述初步处理数据与所述蜂鸣器获取的所述低频信号数据进行对比,消除低频噪音得到用户通话声音数据;
完全背景噪音数据计算模块,用于根据所述用户通话声音数据计算模块得到的所述用户通话声音数据获得完全背景噪音数据。
所述第一麦克风、所述第二麦克风及所述蜂鸣器与所述音频数字信号协处理器连接,所述音频数字信号协处理器连接所述基带应用处理器。
所述根据周围噪音大小自动切换情景模式系统,其中,所述系统还包括音频系统模块,用于提供音频数字和模拟接口,将所述第一麦克风、所述第二麦克风获取的所述外部声音信号数据和所述蜂鸣器获取的所述低频信号数据由模拟信号转换为数字信号,所述音频系统模块分别连接所述第一麦克风、所述第二麦克风及所述蜂鸣器,并与所述音频数字信号协处理器连接。
有益效果
本发明提供一种根据周围噪音大小自动切换情景模式的方法及系统,其通过设置背景噪音与通话声音的对比的技术手段实现了截取纯粹背景噪音的目的,并最终实现移动终端无需手动调节,可根据背景噪音自动切换情景模式的效果,保证了用户无论处于何种环境均具有较佳的通话质量。
附图说明
图1为本发明的根据周围噪音大小自动切换情景模式的方法流程图;
图2为本发明的根据周围噪音大小自动切换情景模式的系统原理框图;
图3为本发明的音频数字信号协处理器的系统原理框图;
图4为本发明较佳实施例中第一、第二麦克风及蜂鸣器的设置位置示意图。
本发明的最佳实施方式
本发明提供一种根据周围噪音大小自动切换情景模式的方法及系统,为使本发明的目的、技术方案及效果更加清楚、明确,以下对本发明进一步详细说明。应当理解,此处所描述的具体实施例仅仅用以解释本发明,并不用于限定本发明。
如图1所示的一种根据周围噪音大小自动切换情景模式的方法,其中,所述方法包括以下步骤:
S100、通过第一麦克风、第二麦克风分别获取外部声音信号数据,通过蜂鸣器获取低频信号数据。
其中,所述外部声音信号数据包括背景噪音信号数据和用户通话声音数据,所述第一麦克风主要用于获取背景噪音信号数据,所述第二麦克风主要用于获取用户通话声音数据。如图4所示,所述第一麦克风110、所述第二麦克风120和所述蜂鸣器130分别设置在移动终端100的不同位置。所述第二麦克风120设置在移动终端靠近用户发声的位置,可以获取大部分的用户通话声音数据,以及小部分的外部环境噪音信号。而所述第一麦克风110设置在移动终端远离用户通话发声的位置,因而,所述第一麦克风110获取大部分的背景噪音信号数据,小部分的用户通话声音数据。所述蜂鸣器130作为传感器,通过其对低频信号的高灵敏度,辅助获取背景噪音中的低频信号噪音数据,例如外部环境中的风声、回声等信号,通过三者所获取的声音信号数据包含了外围环境中的所有声音信号,使其数据最大可能的接近真实的外部环境噪音。较佳的是,所述第一麦克风110和第二麦克风120为微型机电系统麦克风(微型机电系统麦克风是基于微型机电系统技术制造的麦克风,简单的说就是一个电容器集成在微硅晶片上,可以采用表贴工艺进行制造,能够承受很高的回流焊温度,容易与互补金属氧化物半导体工艺及其它音频电路相集成,,并具有改进的噪声消除性能与良好的噪声和电磁辐射抑制性能)。
S200、将步骤S100中所述第一麦克风110和所述第二麦克风120所获取的所述外部声音信号数据进行对比,消除中高频背景噪音得到初步处理数据,将所述初步处理数据与步骤S100中所述蜂鸣器130所获取的所述低频信号数据进行对比,消除低频噪音得到用户通话声音数据,根据所述用户通话声音数据获得完全背景噪音数据。
上述处理过程,是通过预先设置的对获取的所述外部声音信号数据及所述低频信号数据的处理算法进行,该算法所实现的处理过程将背景噪音与用户通话声音最大限度的分离开,相对于传统的只是单纯靠分析人声与背景噪音混合声音的响度的技术来说,本发明的对背景噪音和人声的处理更准确,效果更好。传统靠响度阈值来区分人声和背景噪音的处理过程中,如果通话的时候人声带有情绪,人声的响度大小很可能超过所设置的响度阀值,从而造成系统的误判断。而本发明则没有上述问题,本发明的技术方案按照所设置算法截取纯粹的背景噪音,并利用背景噪音来进一步实现情景模式的切换。
S300、根据步骤S200所得到的完全背景噪音数据切换情景模式。
所述步骤S300具体为:将步骤S200中所述完全背景噪音数据与预先设置的情景模式切换列表进行匹配,应用与所述完全背景噪音数据所匹配对应的情景模式,其中,所述情景模式切换列表中,所述背景噪声分贝值与所述情景模式一一对应。
较佳的是,所述情景模式切换列表中设置第一背景噪音阈值,当完全背景噪音数据高于所述第一背景噪音阈值时,与该完全背景噪音数据所对应的所述情景模式中,所述背景噪音分贝值越大,所述通话音量和所述铃声音量越大。进一步的,所述情景模式切换列表中设置第二背景噪音阈值,当完全背景噪音数据低于所述第二背景噪音阈值时,与该完全背景噪音数据所对应的所述情景模式中,所述背景噪音分贝值越小,所述通话音量和所述铃声音量越小。较佳实施例中,所述第一背景噪音阈值为60分贝(dB)。
由于在60 分贝以上时的背景噪音环境中,人与人之间的交流变得困难,所以设置所述第一背景噪音阈值为60 分贝,如果获取的所述完全背景噪音数据高于所述第一背景噪音阈值,则移动终端进入与该完全背景噪音数据所对应的情景模式,该模式中通话音量及铃声音量都会相应增大,以保证用户能够正常通话,以及保证用户能够听到铃声,不漏掉电话。同理,如果获取的所述完全背景噪音数据低于所述第一背景噪音阈值的情景模式中,它们之间的通话音量及铃声可以相同,但如果所述完全背景噪音数据过小,就需要设置第二背景噪音阈值,当完全背景噪音数据低于所述第二背景噪音阈值时,其所对应的情景模式中的通话音量和铃声会减小,且完全背景噪音数据越低,所对应的所述情景模式的通话音量和铃声就越小,这样能够实现用户在一个很安静的环境中时,自动减小通话声音,保证通话内容不被泄露,另外由于环境很安静,较小的铃音也足够用来提醒用户。
当然,由于人与人之间的听力敏感性不同,用户也可以根据自己听力的敏感性自行设置所述第一背景噪音阈值和所述第二背景噪音阈值,将其设置到合适该用户自己的数值。所述情景模式中包括通话音量大小,铃声音量大小。当然,不同的情景模式也包括其他属性的设置,例如振动的开关,信号指示灯的开关等。
如图2和图3所示的一种设置在移动终端100中具有根据周围噪音大小自动切换情景模式的系统,其中,所述系统包括:第一麦克风110、第二麦克风120、蜂鸣器130、音频数字信号协处理器200、基带应用处理器300。
所述第一麦克风110、第二麦克风120用于获取外部声音信号数据及,所述蜂鸣器130用于获取低频信号数据。
所述音频数字信号协处理器200,用于对所述第一麦克风110、第二麦克风120获取的所述外部声音信号数据及所述蜂鸣器130获取的所述低频信号数据进行分析处理,获得完全背景噪音数据;
所述基带应用处理器300,用于根据所述音频数字信号协处理器200所得到的所述完全背景噪音数据切换情景模式;
所述第一麦克风110、所述第二麦克风120及所述蜂鸣器130与所述音频数字信号协处理器连接200,所述音频数字信号协处理器200连接所述基带应用处理器300;
所述音频数字信号协处理器200包括以下依次连接的模块:
初步处理数据计算模块210、用户通话声音数据计算模块220、完全背景噪音数据计算模块230。
所述初步处理数据计算模块210,用于将所述第一麦克风110和所述第二麦克风120分别所获取的外部声音信号数据进行对比,消除中高频背景噪音得到初步处理数据。
所述用户通话声音数据计算模块220,用于将所述初步处理数据计算模块210获取的所述初步处理数据与所述所述蜂鸣器130获取的所述低频信号数据进行对比,消除低频噪音得到用户通话声音数据。
所述完全背景噪音数据计算模块230,用于根据所述用户通话声音数据计算模块220获取的所述用户通话声音数据获得完全背景噪音数据。
所述系统还包括音频系统模块400,用于提供音频数字和模拟接口,将所述第一麦克风110、所述第二麦克风120获取的所述外部声音信号数据和所述蜂鸣器130获取的所述低频信号数据由模拟信号转换为数字信号。
所述第一麦克风110、所述第二麦克风120及所述蜂鸣器130通过所述音频系统模块400与所述音频数字信号协处理器200连接,所述音频数字信号协处理器200连接所述基带应用处理器300。
较佳实施例中,所述移动终端中设置的有基带处理器600、射频收发器710,射频放大器720。所述基带处理器600分别与所述音频数字信号协处理器200及音频系统模块400连接,所述射频收发器710与所述基带处理器600连接、所述射频放大器720与所述射频收发器710连接,结合本发明的根据周围噪音大小自动切换情景模式的系统形成了一个完整的移动终端系统,使该移动终端系统不仅具有根据周围噪音大小自动切换情景模式的功能,还具有将所提取的纯粹的用户通话声音数据用于通话过程,即相当于为用户通话设置降噪处理,提高了通话质量。
本发明提供的根据周围噪音大小自动切换情景模式的方法及系统,其通过设置背景噪音与通话声音的对比的技术手段实现了截取纯粹背景噪音的目的,并最终实现移动终端无需手动调节,可根据背景噪音自动切换情景模式的效果,保证了用户无论处于何种环境均具有较佳的通话质量。
应当理解的是,本发明的应用不限于上述的举例,对本领域普通技术人员来说,可以根据上述说明加以改进或变换,所有这些改进和变换都应属于本发明所附权利要求的保护范围。
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  1. 一种根据周围噪音大小自动切换情景模式的方法,其中所述方法包括:
    通过第一麦克风、第二麦克风分别获取外部声音信号数据,通过蜂鸣器获取低频信号数据;所述外部声音信号数据包括背景噪音信号数据和用户通话声音数据,所述第一麦克风用于获取背景噪音信号数据,所述第二麦克风用于获取用户通话声音数据;
    将所述第一麦克风和所述第二麦克风所获取的所述外部声音信号数据进行对比,消除中高频背景噪音得到初步处理数据;
    将所述初步处理数据与所述蜂鸣器获取的所述低频信号数据进行对比,消除低频噪音得到用户通话声音数据;
    根据所述用户通话声音数据获得完全背景噪音数据;以及
    将所述完全背景噪音数据与预先设置的情景模式切换列表进行匹配,应用所述完全背景噪音数据所匹配对应的情景模式。
  2. 根据权利要求1所述的根据周围噪音大小自动切换情景模式的方法,其中所述第一麦克风、所述第二麦克风和所述蜂鸣器分别设置在移动终端的不同位置,所述第一麦克风和所述第二麦克风为微型机电系统麦克风。
  3. 根据权利要求1所述的根据周围噪音大小自动切换情景模式的方法,其中所述情景模式切换列表中,所述背景噪音分贝值与所述情景模式一一对应。
  4. 根据权利要求3所述的根据周围噪音大小自动切换情景模式的方法,其中所述情景模式中包括通话音量大小和铃声音量大小。
  5. 根据权利要求4所述的根据周围噪音大小自动切换情景模式的方法,其中所述情景模式切换列表中设置第一背景噪音阈值,当所述完全背景噪音数据高于所述第一背景噪音阈值时,与该完全背景噪音数据所对应的所述情景模式中,所述背景噪音分贝值越大,所述通话音量和所述铃声音量越大。
  6. 根据权利要求5所述的根据周围噪音大小自动切换情景模式的方法,其中所述第一背景噪音阈值为60 分贝。
  7. 根据权利要求4所述的根据周围噪音大小自动切换情景模式的方法,其中所述情景模式切换列表中设置第二背景噪音阈值,当所述完全背景噪音数据低于所述第二背景噪音阈值时,与该完全背景噪音数据所对应的所述情景模式中,所述背景噪音分贝值越小,所述通话音量和所述铃声音量越小。
  8. 一种根据周围噪音大小自动切换情景模式的方法,其中所述方法包括:
    通过第一麦克风、第二麦克风分别获取外部声音信号数据,通过蜂鸣器获取低频信号数据;
    将所述第一麦克风和所述第二麦克风所获取的所述外部声音信号数据进行对比,消除中高频背景噪音得到初步处理数据;
    将所述初步处理数据与所述蜂鸣器获取的所述低频信号数据进行对比,消除低频噪音得到用户通话声音数据;
    根据所述用户通话声音数据获得完全背景噪音数据;以及
    根据所述完全背景噪音数据切换情景模式。
  9. 根据权利要求8所述的根据周围噪音大小自动切换情景模式的方法,其中所述外部声音信号数据包括背景噪音信号数据和用户通话声音数据,所述第一麦克风主要用于获取背景噪音信号数据,所述第二麦克风主要用于获取用户通话声音数据。
  10. 根据权利要求9所述的根据周围噪音大小自动切换情景模式的方法,其中所述第一麦克风、所述第二麦克风和所述蜂鸣器分别设置在所述移动终端的不同位置,所述第一麦克风和所述第二麦克风为微型机电系统麦克风。
  11. 根据权利要求8所述的根据周围噪音大小自动切换情景模式的方法,其中所述根据所述完全背景噪音数据切换情景模式的步骤具体为:将所述完全背景噪音数据与预先设置的情景模式切换列表进行匹配,应用所述完全背景噪音数据所匹配对应的情景模式,其中,所述情景模式切换列表中,所述背景噪音分贝值与所述情景模式一一对应。
  12. 根据权利要求11所述的根据周围噪音大小自动切换情景模式的方法,其中所述情景模式中包括通话音量大小和铃声音量大小。
  13. 根据权利要求12所述的根据周围噪音大小自动切换情景模式的方法,其中所述情景模式切换列表中设置第一背景噪音阈值,当所述完全背景噪音数据高于所述第一背景噪音阈值时,与该完全背景噪音数据所对应的所述情景模式中,所述背景噪音分贝值越大,所述通话音量和铃声音量越大。
  14. 根据权利要求13所述的根据周围噪音大小自动切换情景模式的方法,其中所述第一背景噪音阈值为60 分贝。
  15. 根据权利要求12所述的根据周围噪音大小自动切换情景模式的方法,其中所述情景模式切换列表中设置第二背景噪音阈值,当所述背景噪音分贝值低于所述第二背景噪音阈值时,与该背景噪音分贝值所对应的所述情景模式中,所述背景噪音分贝值越小,所述通话音量和所述铃声音量越小。
  16. 一种根据周围噪音大小自动切换情景模式系统,其中所述系统包括:
    第一麦克风、第二麦克风,用于获取外部声音信号数据;
    蜂鸣器,用于获取低频信号数据;
    音频数字信号协处理器,用于对所述第一麦克风、所述第二麦克风获取的所述声音信号数据及所述蜂鸣器获取的所述低频信号数据进行分析处理,获得完全背景噪音数据;
    基带应用处理器,用于根据所述音频数字信号协处理器得到的所述完全背景噪音数据切换情景模式;
    所述第一麦克风、所述第二麦克风及所述蜂鸣器与所述音频数字信号协处理器连接,所述音频数字信号协处理器连接所述基带应用处理器;
    所述音频数字信号协处理器包括以下依次连接的模块:
    初步处理数据计算模块,用于将所述第一麦克风和所述第二麦克风所获取的外部声音信号数据进行对比,消除中高频背景噪音得到初步处理数据;
    用户通话声音数据计算模块,用于将所述初步处理数据计算模块得到的所述初步处理数据与所述蜂鸣器获取的所述低频信号数据进行对比,消除低频噪音得到用户通话声音数据;
    完全背景噪音数据计算模块,用于根据所述用户通话声音数据计算模块得到的所述用户通话声音数据获得完全背景噪音数据。
  17. 根据权利要求16所述的根据周围噪音大小自动切换情景模式系统,其中所述系统还包括:
    音频系统模块,用于提供音频数字和模拟接口,将所述第一麦克风、所述第二麦克风获取的所述外部声音信号数据和所述蜂鸣器获取的所述低频信号数据由模拟信号转换为数字信号;
    所述音频系统模块分别连接所述第一麦克风、所述第二麦克风及所述蜂鸣器,并与所述音频数字信号协处理器连接。
  18. 根据权利要求17所述的根据周围噪音大小自动切换情景模式系统,其中所述第一麦克风、所述第二麦克风和所述蜂鸣器分别设置在移动终端的不同位置,所述第一麦克风和第二麦克风为微型机电系统麦克风。
  19. 根据权利要求16所述的根据周围噪音大小自动切换情景模式系统,其中所述基带应用处理器,用于根据所述音频数字信号协处理器所得到的所述完全背景噪音数据与预先设置的情景模式切换列表进行匹配,应用所述完全背景噪音数据所匹配对应的情景模式,其中,所述情景模式切换列表中,所述背景噪声分贝值与所述情景模式一一对应。
  20. 根据权利要求16所述的根据周围噪音大小自动切换情景模式系统,其中所述情景模式中包括通话音量大小和铃声音量大小。
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