WO2012065392A1 - Method and device for processing the quality of voice call - Google Patents

Method and device for processing the quality of voice call Download PDF

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Publication number
WO2012065392A1
WO2012065392A1 PCT/CN2011/071792 CN2011071792W WO2012065392A1 WO 2012065392 A1 WO2012065392 A1 WO 2012065392A1 CN 2011071792 W CN2011071792 W CN 2011071792W WO 2012065392 A1 WO2012065392 A1 WO 2012065392A1
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WIPO (PCT)
Prior art keywords
parameters
audio
mobile terminal
scene
user
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PCT/CN2011/071792
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French (fr)
Chinese (zh)
Inventor
魏国华
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中兴通讯股份有限公司
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Publication of WO2012065392A1 publication Critical patent/WO2012065392A1/en

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Classifications

    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04MTELEPHONIC COMMUNICATION
    • H04M1/00Substation equipment, e.g. for use by subscribers
    • H04M1/60Substation equipment, e.g. for use by subscribers including speech amplifiers
    • 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 communications, and in particular to a method and apparatus for processing voice call quality.
  • a mobile terminal for example, a mobile phone or the like
  • the following processing methods are currently used:
  • the mobile terminal can use the dual microphone noise reduction technology, but this method increases the hardware design cost.
  • this dual-mike noise reduction technology completely filters out the environmental unsteady noise, so that the called end can not truly feel the real environment where the caller is located.
  • the mobile terminal can also dynamically adjust the receiving gain and the transmission gain of the mobile terminal to improve the voice call quality only by detecting the magnitude of the ambient noise by a microphone (MIC) on the mobile terminal motherboard.
  • MIC microphone
  • a primary object of the present invention is to provide a method and apparatus for processing voice call quality to at least solve the above problems.
  • a method for processing voice call quality including: setting a plurality of scenarios in a mobile terminal, wherein each scenario corresponds to a set of audio parameters;
  • the audio parameters corresponding to a scene are set to the mobile terminal.
  • the audio parameter corresponding to the scene is determined according to at least one of the following: an age group of the user, a gender of the user, and an environment in which the mobile terminal is in a call.
  • the following steps are used to determine audio parameters corresponding to the scene: collecting audio samples corresponding to the scene; using the audio samples to perform testing in a standard anechoic chamber; determining the scene according to the test result Corresponding audio parameters.
  • the audio parameter comprises at least one of: a parameter for controlling the magnitude of the sound gain in the transmitting and/or receiving direction, a parameter for adjusting the digital gain on the transmitting and/or receiving channel, adjusting the transmitting and/or receiving channel.
  • setting the mobile terminal by using an audio parameter corresponding to one of the multiple scenarios includes: writing audio parameters corresponding to one of the multiple scenarios to digital signal processing In the DSP register, the terminal sets the mobile terminal according to an audio parameter in the DSP register.
  • a processing device for voice call quality which is located in a mobile terminal, and includes: a first setting module, configured to set a plurality of scenarios, wherein each scenario corresponds to a set of audio parameters;
  • the second setting module is configured to set the mobile terminal by using an audio parameter corresponding to one of the multiple scenarios.
  • the device further includes: a parameter preset module, configured to determine an audio parameter corresponding to the scene according to at least one of the following: an age group of the user, a gender of the user, and the mobile terminal is in a call surroundings.
  • the device further includes: a collection module configured to collect audio samples corresponding to the scene; a test module configured to perform testing in the standard anechoic chamber using the audio sample; and determining a module, configured to The result of the test determines the audio parameters corresponding to the scene.
  • the audio parameter corresponding to the scene set by the first setting module includes at least one of the following: a parameter for controlling a size of a sound gain in a transmitting and/or receiving direction, and a parameter for adjusting a digital gain on a transmitting and/or receiving channel.
  • the device further includes: a writing module, configured to write audio parameters corresponding to one of the plurality of scenarios into a digital signal processing DSP register, wherein the terminal is configured according to the DSP register
  • the audio parameters are set for the mobile terminal.
  • FIG. 1 is a flow chart showing a method for processing voice call quality according to an embodiment of the present invention
  • FIG. 2 is a block diagram showing a structure of a voice call quality processing device according to an embodiment of the present invention
  • BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1
  • FIG. 4 is a schematic diagram of an environment used by a user body-subject subjective test according to an embodiment of the present invention
  • FIG. 5 is an audio diagram of a mobile terminal according to an embodiment of the present invention
  • Module Block Diagram is an audio diagram of a mobile terminal according to an embodiment of the present invention
  • Figure 6 is a schematic diagram of scene relationships based on gender, age, and location of use, in accordance with an embodiment of the present invention.
  • the method includes the following steps: Step S102: Set a plurality of scenarios in a mobile terminal, where each scenario corresponds to a set of audio parameters; Step S104: Set the mobile terminal by using audio parameters corresponding to one of the plurality of scenarios.
  • Step S102 Set a plurality of scenarios in a mobile terminal, where each scenario corresponds to a set of audio parameters; Step S104: Set the mobile terminal by using audio parameters corresponding to one of the plurality of scenarios.
  • the spectrum of the auditory and call speech is used to determine the listening ability of different users, and then to set the appropriate scene. For example, at least one of the following determines the audio parameters corresponding to the scene: the age of the user, the gender of the user, and the environment in which the mobile terminal is in a call.
  • the user can select a separate voice call configuration according to his or her age, gender, and different usage environments in which the voice call is made, which can give the user more voice call quality selection options. The user significantly improves the quality of voice calls without replacing the mobile terminal.
  • a preferred method for determining an audio parameter corresponding to a scene is also provided (of course, the method for determining an audio parameter may not be used, but the determining method is relatively easy to implement), and the method includes the following Step ⁇ 1 Step S1, collecting the audio samples corresponding to the scene; Step S2, using the audio samples to test in the standard anechoic chamber; Step S3, determining the audio parameters corresponding to the scene according to the test results.
  • the audio parameters comprise at least one of: parameters for controlling the magnitude of the sound gain in the transmit and/or receive directions, adjusting parameters of the digital gain on the transmit and/or receive channels, adjusting the analog gain on the transmit and/or receive channels
  • the audio parameters corresponding to one of the multiple scenes can be written into a digital signal processing (DSP) register, and the terminal can only use the DSP register.
  • DSP digital signal processing
  • FIG. 2 is a structural block diagram of a processing apparatus for voice call quality according to an embodiment of the present invention. The device is used to implement the foregoing embodiment and its preferred embodiments.
  • the device includes: a first setting module 10 and a second setting module 20.
  • the structure will be described below.
  • the first setting module 10 is configured to set a plurality of scenarios in the mobile terminal, where each scenario corresponds to a set of audio parameters; and the second setting module 20 is configured to use audio corresponding to one of the multiple scenarios.
  • the parameters are set for the mobile terminal.
  • 3 is a structural block diagram of a processing apparatus for a preferred voice call quality according to an embodiment of the present invention. As shown in FIG.
  • the apparatus further includes: a parameter preset module 302, configured to determine an audio parameter corresponding to the scene according to at least one of the following: : The age of the user, the gender of the user, and the environment in which the mobile terminal is talking.
  • the device further includes: a collection module 304 configured to collect audio samples corresponding to the scene; a test module 306 configured to perform testing in a standard anechoic chamber using the audio samples; and the determining module 308 is configured to The result of the test determines the audio parameters corresponding to the scene.
  • the audio parameter corresponding to the scene set by the first setting module 10 includes at least one of the following: a parameter for controlling the size of the sound gain in the transmitting and/or receiving direction, a parameter for adjusting the digital gain on the transmitting and/or receiving channel, Adjust the parameters of the analog gain on the transmit and/or receive channels, the parameters that modulate the frequency of the transmitted and/or received speech, the parameters that suppress the background noise transmission, and the parameters that enhance the double talk effect.
  • the device further includes: a writing module 310, configured to write audio parameters corresponding to one of the plurality of scenes into the DSP register, and the terminal sets the mobile terminal according to the audio parameters in the DSP register.
  • call usage scenarios are built in the built-in programming software version of the mobile terminal, such as selecting the user's age level, gender, and the real environment in which the user voice calls: family, conference room, road, Beach, bus, etc. Users can set according to their own call needs and actual usage. Through the above three modes, users can choose the voice call quality mode that suits them.
  • separate different audio parameters are provided in the built-in software of the mobile terminal, so that the user can write the DSP register in real time when the user selects the scene, thereby achieving the purpose of improving the quality of the voice call.
  • the individual audio parameters of different scenes are obtained by capturing different sound samples, for example, sound samples of different age levels, sound samples of different genders, sound samples of different places of use, and in standard
  • the anechoic chamber passes the test system.
  • the Advanced Communication Quality Analysis (ACQUA) audio test system can be used to test the terminal, adjust the audio parameters in real time, and process the test results, for example, the average subjective score ( The Mean Opinion Score, referred to as the MOS) score size or the ITU-defined voice test standard, determines the audio parameters that use the best results.
  • ACQUA Advanced Communication Quality Analysis
  • FIG. 4 is a schematic diagram of an environment used by a user body-subject subjective test according to an embodiment of the present invention.
  • the voice call quality of the mobile terminal in the sending direction can be tested. If the mobile terminal and the fixed telephone in the two rooms are interchanged, the quality of the voice call in the direction in which the mobile terminal is received can be tested.
  • the background noise of the real scene is simulated in the room through the speaker, so as to test the voice call quality of the mobile terminal in different scenarios.
  • FIG. 5 is a structural block diagram of an audio module of a mobile terminal according to an embodiment of the present invention.
  • algorithms for generally adjusting voice call quality include Auto Gain Control (AGC) module, digital gain, analog gain, and Finite Impulse Response (FIR).
  • AGC Auto Gain Control
  • FIR Finite Impulse Response
  • IIR Filter and Echo Canceller no P-monthly response (Infinite Impulse Response, referred to as IIR) Filter and Echo Canceller (EC) modules.
  • the AGC module is used to control the size of the sound gain in the transmitting and receiving directions, to avoid the excessive or too small sound affecting the user's subjective hearing experience. It is based on the set compression threshold, extended threshold, compression slope, extended slope and static gain registers. Gain adjustment is performed and low frequency noise can be filtered out. Both digital gain and analog gain can increase and decrease the gain on the transmit or receive channels.
  • the FIR or IIR filter is used to modulate the frequency response of the received or transmitted speech, and can be adjusted to achieve optimal conditions according to different scenarios.
  • the EC module is responsible for eliminating echoes during the mobile terminal's call, and the registers of this module can suppress the transmission of background noise and enhance the double talk effect.
  • the above DSP registers need to be adjusted in different scenarios. The following description will be based on the spectrum range in which the sounds are different.
  • 60 ⁇ 100Hz This frequency affects the richness of the sound and is the pitch area of the bass. If the frequency is very full, the tone will look thick and thick. If the frequency is insufficient, the tone will become weak; if the frequency is too strong, the tone will have a low frequency resonance and a sensation.
  • 100 ⁇ 150Hz This frequency affects the fullness of the tone. If this frequency component is enhanced, it will create a sense of space and thickness in the room. If this frequency component is missing, the tone will become thin. Pale; if this frequency component is too strong, the tone will appear turbid and the clarity of the voice will deteriorate. 150 ⁇ 300Hz: This frequency affects the strength of the sound, especially the strength of the male voice. This frequency is the low-frequency fundamental frequency of the male voice, and is also the root audio frequency of the chord in the tone. If this frequency component is lacking, the tone will appear soft and fluttering, and the voice will become soft. If this frequency component is too strong, the sound will become stiff and unnatural, and there is no special feature.
  • the mid-band frequency determines the sound intensity. If the sound exceeds +5dB ⁇ 10dB, the sound becomes blurred, the sharpness decreases, and the drop is -6 ⁇ 10dB. The sound lacks strength and is thin, and the sound is hard and narrow.
  • 300 ⁇ 500Hz This frequency is the main zone frequency of the voice. The frequency of this frequency is full and the voice is strong. If the amplitude of this frequency is insufficient, the sound will appear hollow and not solid; if the frequency is too strong, the tone will become monotonous, the relative frequency component will be less, the high frequency will be less, and the voice will become similar to the phone. The sound of the sound is the same, it looks very monotonous.
  • 500 ⁇ lKHz The frequency is the pitch frequency area of the human voice and is an important frequency range. If this frequency is full, the vocal contour is clear and the overall feeling is good; if the frequency is not enough, the voice will have a sense of contraction; if the frequency is too strong, the voice will have a feeling of forward accentuation.
  • the voice produces an auditory feeling that enters the person in advance.
  • 800Hz This frequency amplitude affects the strength of the tone. If this frequency is full, the tone will appear strong and powerful; if this frequency is insufficient, the tone will appear slack, that is, the characteristic characteristics below 800 Hz will be prominent, and the low frequency component will be obvious; if this frequency is too high, the throat will be produced. Sound sense. People has a throat. People has a certain throat sound. If there are too many throat sounds in the tone, the sound will be lost. l ⁇ 2KHz: This frequency range has obvious transparency and smoothness. If this frequency is lacking, the tone is loose and the tone is out of line; if this frequency is too strong, the tone has a jump.
  • 2 ⁇ 3KHz This frequency is the most sensitive frequency band that affects the brightness of the sound. If the frequency component is rich, the brightness of the tone will be enhanced. If the frequency is insufficient, the tone will become awkward; If this frequency component is too strong, the tone will appear dull, hard and unnatural.
  • the medium to high frequency band of l ⁇ 3KHz plays an important role in brightness, sharpness and presence. If the frequency band exceeds +3 ⁇ 5dB, the sound will be hardened. If it exceeds +5 ⁇ 10dB, metal sound will appear, and the drop will be -3 ⁇ 5dB. Hardening, more than +5 ⁇ 10dB will appear metal sound, falling -3 ⁇ 5dB will make the tone lose brightness, falling -5 ⁇ 10dB The sound is boring, not clear.
  • 3 ⁇ 4KHz The penetration of this frequency is very strong.
  • the resonant frequency of the human ear cavity is 1 to 4 ⁇ , so the human ear is also very sensitive to this frequency. If this frequency component is too small, the hearing ability will be worse, and the voice will be blurred. If this frequency component is too strong, it will produce a coughing sensation.
  • the low frequency of the general male voice spectrum is relatively rich, resulting in the other side of the mobile terminal (ie, the called end), the sound will be boring, in this case, you can pass the FIR Or the IIR filter performs the adjustment of the chirp frequency, and raises the spectral gain between the 100-500HZ portion of the chirp frequency to improve the call quality.
  • the high frequency of the female voice spectrum is rich, which causes the other side (called end) of the mobile terminal to sound sharp and harsh, and can use the FIR or IIR filter to obtain the spectral gain between 3000-4000HZ at high frequencies. Perform a certain pressure to improve the quality of the call.
  • the age level of people is generally divided into old age, middle age, youth, and children.
  • the distribution of sound spectrum is different, and it is necessary to adjust with different FIR and IIR filter parameters. . If the elderly have poor hearing, you need to improve the receiving channel.
  • HAC Hear Assist Carrier
  • the noise suppression algorithm can be used to suppress the vehicle noise to a greater extent, and can also be in the AGC pair.
  • Figure 6 shows a scenario linkage diagram based on gender, age, and location of use. Based on Figure 6, there are about 40 usage scenarios that can be provided, and can be added according to actual needs. Based on the gender, age level, and location of the mobile terminal user described above, approximately 40 audio call scene configurations are required. Each scene configuration is rigorously tested by the audio lab using different audio samples, and the actual scene is tested to ensure the voice call quality of each scene. When the user makes no choices, the mobile terminal is configured with a set of default audio parameters. When the user selects his or her gender, age level, and the use environment of the call according to his/her own situation, the mobile terminal invokes the specified scene configuration through the setting of the user, thereby achieving the purpose of improving the quality of the voice call, and greatly improving the mobile terminal.
  • the flexibility to improve the quality of voice calls provides users with great convenience.
  • the requirements of the user for the quality of the voice call can be satisfied, and the cost of the mobile terminal manufacturer is reduced, and the quality of the voice call is significantly improved when the user uses the mobile terminal.
  • the above 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.
  • 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.

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Environmental & Geological Engineering (AREA)
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  • Computer Networks & Wireless Communication (AREA)
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Abstract

A method and device for processing the quality of a voice call are disclosed in the present invention. The method includes: setting multiple scenes in a mobile terminal, wherein each scene corresponding to a set of audio parameters; using the audio parameters corresponding to one scene of the multiple scenes to set the mobile terminal. With the invention, the problem that the quality of a voice call can not meet the actual needs of a user in the prior art is solved, and the user experience is improved.

Description

语音通话质量的处理方法及装置 技术领域 本发明涉及通信领域, 具体而言, 涉及一种语音通话质量的处理方法及 装置。 背景技术 目前, 在移动终端 (例如, 手机等) 上, 为了提高通话的语音质量, 目 前釆用以下的处理方式: 移动终端可以釆用双麦克降噪技术, 但是这种方法增加了硬件设计成本 以及增加了结构设计的复杂性, 这种双麦克降噪技术对于环境非稳态噪声进 行完全的滤除使得被叫端无法真实感受呼叫端所处的真实环境。 移动终端还可以仅仅通过移动终端主板上的麦克 ( MIC )检测环境噪声 的大小来动态调节移动终端的接收增益和发送增益来改善语音通话质量。 由 于仅仅通过 MIC釆集的环境噪声无法判断用户所处的确切的噪声场所,提高 发送增益和接收增益并能不能真正解决语音通话质量问题, 反而带来更多的 语音失真, 影响用户主观感受。 上述两种处理方法均是在移动终端中内置固定软件版本, 这样的处理方 式, 在出厂后用户无法进行爹改, 导致语音通话质量不能满足用户的真实需 求, 在这种情况下, 用户可能选择退机或者去售后更新软件版本, 从而影响 了用户体险。 发明内容 本发明的主要目的在于提供一种语音通话质量的处理方法及装置, 以至 少解决上述问题。 才艮据本发明的一个方面, 提供了一种语音通话质量的处理方法, 包括: 在移动终端中设置多种场景, 其中, 每一种场景对应一套音频参数; 使用所 述多种场景中的一种场景所对应的音频参数对所述移动终端进行设置。 优选地, 才艮据以下至少之一确定所述场景所对应的音频参数: 用户的年 龄段、 用户的性别、 所述移动终端在通话时所处的环境。 优选地, 釆用以下步骤确定所述场景对应的音频参数: 釆集所述场景对 应的音频样本; 使用所述音频样本在标准的消声室进行测试; 才艮据测试的结 果确定所述场景对应的音频参数。 优选地, 所述音频参数包括以下至少之一: 用于控制发送和 /或接收方向 上声音增益大小的参数、 调整发送和 /或接收通道上数字增益的参数、 调整发 送和 /或接收通道上模拟增益的参数、 调制发送和 /或接收语音的频率的参数、 抑制背景噪声传输的参数、 增强双讲通话效果的参数。 优选地, 使用所述多种场景中的一种场景所对应的音频参数对所述移动 终端进行设置包括: 将所述多种场景中的一种场景所对应的音频参数写入到 数字信号处理 DSP寄存器中, 所述终端根据所述 DSP寄存器中的音频参数 对所述移动终端进行设置。 根据本发明的另一方面, 提供了一种语音通话质量的处理装置, 位于移 动终端中, 包括: 第一设置模块, 设置为设置多种场景, 其中, 每一种场景 对应一套音频参数; 第二设置模块, 设置为使用所述多种场景中的一种场景 所对应的音频参数对所述移动终端进行设置。 优选地, 所述装置还包括: 参数预置模块, 设置为根据以下至少之一确 定所述场景所对应的音频参数: 用户的年龄段、 用户的性别、 所述移动终端 在通话时所处的环境。 优选地, 所述装置还包括: 釆集模块, 设置为釆集所述场景对应的音频 样本; 测试模块, 设置为使用所述音频样本在标准的消声室进行测试; 确定 模块, 设置为根据测试的结果确定所述场景对应的音频参数。 优选地, 所述第一设置模块设置的场景对应的音频参数包括以下至少之 一: 用于控制发送和 /或接收方向上声音增益大小的参数、 调整发送和 /或接 收通道上数字增益的参数、 调整发送和 /或接收通道上模拟增益的参数、 调制 发送和 /或接收语音的频率的参数、 抑制背景噪声传输的参数、 增强双讲通话 效果的参数。 优选地, 所述装置还包括: 写入模块, 设置为将所述多种场景中的一种 场景所对应的音频参数写入到数字信号处理 DSP寄存器中,所述终端根据所 述 DSP寄存器中的音频参数对所述移动终端进行设置。 通过本发明, 解决了现有技术中导致语音通话质量不能满足用户的真实 需求的问题, 提高了用户体验。 附图说明 此处所说明的附图用来提供对本发明的进一步理解, 构成本申请的一部 分, 本发明的示意性实施例及其说明用于解释本发明, 并不构成对本发明的 不当限定。 在附图中: 图 1是才艮据本发明实施例的一种语音通话质量的处理方法流程图; 图 2是 居本发明实施例的语音通话质量的处理装置结构框图; 图 3是 居本发明实施例优选的语音通话质量的处理装置结构框图; 图 4是根据本发明实施例的用户体-险者主观测试时所使用的环境示意 图; 图 5是 居本发明实施例的移动终端的音频模块结构框图; 图 6是根据本发明实施例的基于性别, 年龄以及使用场所的场景联系示 意图。 具体实施方式 下文中将参考附图并结合实施例来详细说明本发明。 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组合。 图 1是 居本发明实施例的一种语音通话质量的处理方法流程图。 如图 1所示, 该方法包括如下步骤: 步骤 S 102, 在移动终端中设置多种场景, 其中, 每一种场景对应一套音 频参数; 步骤 S 104,使用多种场景中的一种场景所对应的音频参数对移动终端进 行设置。 通过上述步骤, 移动终端中设置了多种场景, 从而为用户提供了多种选 择,使用户可以根据自己需求选择合适的场景, 来获得满意的语音通话质量。 现有的移动终端对于用户的性别, 年龄层次以及移动终端使用场所无法 进行判断, 在本实施例的下面的优选实施例中根据心理声学和生理声学、 例 如, 可以根据不同性别或年龄层次的用户的听觉和通话语音的频谱范围来确 定不同用户的听力能力, 进而设置合适的场景。 例如, 居以下至少之一确定场景所对应的音频参数: 用户的年龄段、 用户的性别、 移动终端在通话时所处的环境。 在该优选实施方式下, 用户可 以根据自己的年龄, 性别以及进行语音通话时所处的不同使用环境来选择一 种单独的语音通话配置, 这样可以给予用户更多的语音通话质量选择方案, 方便用户在不更换移动终端的前提下显著提升语音通话质量。 在本实施例中,还提供了一种优选的确定场景对应的音频参数的方法(当 然也可以不釆用这种确定音频参数的方法, 不过这种确定方法比较容易实 现), 该方法包括如下步^^ 步骤 S 1 , 釆集场景对应的音频样本; 步骤 S2 , 使用音频样本在标准的消声室进行测试; 步骤 S3 , 才艮据测试的结果确定场景对应的音频参数。 优选地, 音频参数包括以下至少之一: 用于控制发送和 /或接收方向上声 音增益大小的参数、 调整发送和 /或接收通道上数字增益的参数、 调整发送和 /或接收通道上模拟增益的参数、 调制发送和 /或接收语音的频率的参数、 抑 制背景噪声传输的参数、 增强双讲通话效果的参数。 当然, 为了更加便于在移动终端中实现, 可以将多种场景中的一种场景 所对应的音频参数写入到数字信号处理 (Digital Signal Processing, 简称为 DSP ) 寄存器中, 终端才艮据 DSP寄存器中的音频参数对移动终端进行设置。 通过上述实施例及其优选实施方式, 在移动终端中预置了几种移动终端 使用场景模式, 使得用户在通话过程中可以根据自己的性别, 年龄层次, 以 及进行语音通话时所处的位置和环境, 对场景进行选择, 然后移动终端才艮据 用户所选择场景配置动态地写入移动终端的 DSP中基于不同场景定制的调 节好的音频参数, 这样很好地满足了用户对语音通话质量提出的较高要求, 降低了移动终端厂商售后成本, 也使得用户使用移动终端时, 语音通话质量 有了显著的提升。 图 2是 居本发明实施例的语音通话质量的处理装置结构框图, 该装置 用于实现上述实施例及其优选实施方式, 已经进行过说明的不再赘述, 下面 对该结构中涉及到的模块进行说明, 如图 2所示, 该装置包括: 第一设置模 块 10、 第二设置模块 20, 下面对该结构进行说明。 第一设置模块 10 , 设置为在移动终端中设置多种场景, 其中, 每一种场 景对应一套音频参数; 第二设置模块 20, 设置为使用多种场景中的一种场景 所对应的音频参数对移动终端进行设置。 图 3是 居本发明实施例优选的语音通话质量的处理装置结构框图, 如 图 3所示, 该装置还包括: 参数预置模块 302 , 设置为根据以下至少之一确 定场景所对应的音频参数: 用户的年龄段、 用户的性别、 移动终端在通话时 所处的环境。 优选地, 该装置还包括: 釆集模块 304 , 设置为釆集场景对应的音频样 本; 测试模块 306 , 设置为使用音频样本在标准的消声室进行测试; 确定模 块 308 , 设置为才艮据测试的结果确定场景对应的音频参数。 优选地, 第一设置模块 10设置的场景对应的音频参数包括以下至少之 一: 用于控制发送和 /或接收方向上声音增益大小的参数、 调整发送和 /或接 收通道上数字增益的参数、 调整发送和 /或接收通道上模拟增益的参数、 调制 发送和 /或接收语音的频率的参数、 抑制背景噪声传输的参数、 增强双讲通话 效果的参数。 优选地, 该装置还包括: 写入模块 310, 设置为将多种场景中的一种场 景所对应的音频参数写入到 DSP寄存器中, 终端根据 DSP寄存器中的音频 参数对移动终端进行设置。 下面结合另一个优选实施例进行说明, 该优选实施例结合上述实施例及 其优选实施方式。 本优选实施例, 在移动终端出厂的内置烧录软件版本中内置几种通话使 用场景, 如选择用户的年龄层次、 性别, 以及用户语音通话时所处的真实环 境: 家庭, 会议室, 马路, 海滩, 公交等。 用户可以根据自己的通话需求和 实际使用情况进行设置, 通过如上三种模式的选择, 用户可以选择适合自己 的语音通话质量模式。 在本优选实施例中, 针对每种场景, 在移动终端内置 软件中提供单独的不同的音频参数,供用户选择好场景时实时地写入 DSP寄 存器, 达到提升语音通话质量的目的。 在本优选实施例, 不同场景的单独的音频参数的获得是通过釆取不同的 声音样本, 例如, 不同年龄层次的声音样本, 不同性别的声音样本, 不同使 用场所的声音样本, 并在标准的消声室通过测试系统, 例如可以使用先进通 信质量分析 ( Advanced Communication Quality Analysis, 简称为 ACQUA)音 频测试系统, 对终端进行测试, 实时调节音频参数, 居测试结果进行处理, 例如, 平均主观分数 ( Mean Opinion Score, 简称为 MOS )分值的大小或 ITU 规定的语音测试标准决定釆用最佳效果的音频参数。 然后通过用户体验志愿 者主观听觉感受得出的 MOS分来判断此时的语音通话质量是否是处于一个 较好的状态, 最终确定选择的音频参数是否需要继续调整。 图 4是根据本发明实施例的用户体-险者主观测试时所使用的环境示意 图, 如图 4所示, 可以测试移动终端发送方向的语音通话质量。 如果将两个 房间中的移动终端和固定电话进行互换, 则可以测试移动终端接收方向的语 音通话质量。 另外针对不同使用环境的噪声, 通过扬声器在房间内发出模拟 真实场景的背景噪声, 这样来测试不同场景下移动终端的语音通话质量。 在本实施例中, 还可以通过调整的音频参数通过写入 DSP寄存器生效, 使用的 DSP寄存器以及相应算法包括滤波器, 模拟增益, 数字增益, 回声算 法等。 下面结合移动终端中的与音频相关的模块进行说明。 需要说明的是, 以 下的模块仅仅是示例性说明, 但这些功能的实现并不限于在以下的模块中实 现。 图 5是根据本发明实施例的移动终端的音频模块结构框图。如图 5所示, 针对移动终端, 一般调节语音通话质量的算法包括自动增益控制 ( Auto Gain Control,简称为 AGC)模块,数字增益,模拟增益,有限脉冲响应( Finite Impulse Response, 简称为 FIR)或无 P艮月永冲响应 ( Infinite Impulse Response, 简称为 IIR)滤波器和回声消除 ( Echo Canceller, 简称为 EC )模块等。 其中, AGC 模块用来控制发送和接收方向上声音增益的大小, 避免声音过大或过小影响 用户主观听觉感受, 它根据设置的压缩门限、 扩展门限、 压缩斜率、 扩展斜 率和静态增益等寄存器进行增益调整, 并且可以滤除低频噪声。 数字增益和 模拟增益都可以提高和减小发送或接收通道上的增益。 FIR或 IIR滤波器用 来对接收或发送语音的频率响应进行调制, 可以根据不同的场景进行调整, 使其达到最佳状态。 EC模块负责消除移动终端通话过程中的回声, 而且通 过此模块的寄存器可以抑制背景噪声的传输以及增强双讲( double talk )的通 话效果。 对于不同的语音通话质量模式或场景,上述的 DSP寄存器需要 居不同 的场景进行调节。 下面根据声音不同的频谱范围进行说明。 The present invention relates to the field of communications, and in particular to a method and apparatus for processing voice call quality. BACKGROUND At present, in a mobile terminal (for example, a mobile phone or the like), in order to improve the voice quality of a call, the following processing methods are currently used: The mobile terminal can use the dual microphone noise reduction technology, but this method increases the hardware design cost. As well as increasing the complexity of the structural design, this dual-mike noise reduction technology completely filters out the environmental unsteady noise, so that the called end can not truly feel the real environment where the caller is located. The mobile terminal can also dynamically adjust the receiving gain and the transmission gain of the mobile terminal to improve the voice call quality only by detecting the magnitude of the ambient noise by a microphone (MIC) on the mobile terminal motherboard. Since the ambient noise of the MIC is not able to determine the exact noise location where the user is located, improving the transmission gain and the receiving gain can not really solve the problem of voice call quality, but brings more speech distortion and affects the subjective feeling of the user. The above two processing methods are built-in fixed software versions in the mobile terminal. Such a processing method cannot be tampered with by the user after leaving the factory, and the quality of the voice call cannot meet the real needs of the user. In this case, the user may select Retire the machine or update the software version after the sale, which affects the user's physical insurance. SUMMARY OF THE INVENTION A primary object of the present invention is to provide a method and apparatus for processing voice call quality to at least solve the above problems. According to an aspect of the present invention, a method for processing voice call quality is provided, including: setting a plurality of scenarios in a mobile terminal, wherein each scenario corresponds to a set of audio parameters; The audio parameters corresponding to a scene are set to the mobile terminal. Preferably, the audio parameter corresponding to the scene is determined according to at least one of the following: an age group of the user, a gender of the user, and an environment in which the mobile terminal is in a call. Preferably, the following steps are used to determine audio parameters corresponding to the scene: collecting audio samples corresponding to the scene; using the audio samples to perform testing in a standard anechoic chamber; determining the scene according to the test result Corresponding audio parameters. Preferably, the audio parameter comprises at least one of: a parameter for controlling the magnitude of the sound gain in the transmitting and/or receiving direction, a parameter for adjusting the digital gain on the transmitting and/or receiving channel, adjusting the transmitting and/or receiving channel. The parameters of the analog gain, the parameters of the frequency at which the transmitted and/or received speech are modulated, the parameters that suppress the transmission of the background noise, and the parameters that enhance the effect of the double talk. Preferably, setting the mobile terminal by using an audio parameter corresponding to one of the multiple scenarios includes: writing audio parameters corresponding to one of the multiple scenarios to digital signal processing In the DSP register, the terminal sets the mobile terminal according to an audio parameter in the DSP register. According to another aspect of the present invention, a processing device for voice call quality is provided, which is located in a mobile terminal, and includes: a first setting module, configured to set a plurality of scenarios, wherein each scenario corresponds to a set of audio parameters; The second setting module is configured to set the mobile terminal by using an audio parameter corresponding to one of the multiple scenarios. Preferably, the device further includes: a parameter preset module, configured to determine an audio parameter corresponding to the scene according to at least one of the following: an age group of the user, a gender of the user, and the mobile terminal is in a call surroundings. Preferably, the device further includes: a collection module configured to collect audio samples corresponding to the scene; a test module configured to perform testing in the standard anechoic chamber using the audio sample; and determining a module, configured to The result of the test determines the audio parameters corresponding to the scene. Preferably, the audio parameter corresponding to the scene set by the first setting module includes at least one of the following: a parameter for controlling a size of a sound gain in a transmitting and/or receiving direction, and a parameter for adjusting a digital gain on a transmitting and/or receiving channel. Adjusting the parameters of the analog gain on the transmit and/or receive channels, the parameters of the frequency at which the transmitted and/or received speech are modulated, the parameters that suppress the transmission of the background noise, and the parameters that enhance the effect of the double talk. Preferably, the device further includes: a writing module, configured to write audio parameters corresponding to one of the plurality of scenarios into a digital signal processing DSP register, wherein the terminal is configured according to the DSP register The audio parameters are set for the mobile terminal. The invention solves the problem that the quality of the voice call cannot meet the real needs of the user in the prior art, and improves the user experience. BRIEF DESCRIPTION OF THE DRAWINGS The accompanying drawings, which are set to illustrate,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,,, BRIEF DESCRIPTION OF THE DRAWINGS FIG. 1 is a flow chart showing a method for processing voice call quality according to an embodiment of the present invention; FIG. 2 is a block diagram showing a structure of a voice call quality processing device according to an embodiment of the present invention; BRIEF DESCRIPTION OF THE DRAWINGS FIG. 4 is a schematic diagram of an environment used by a user body-subject subjective test according to an embodiment of the present invention; FIG. 5 is an audio diagram of a mobile terminal according to an embodiment of the present invention; Module Block Diagram; Figure 6 is a schematic diagram of scene relationships based on gender, age, and location of use, in accordance with an embodiment of the present invention. BEST MODE FOR CARRYING OUT THE INVENTION Hereinafter, the present invention will be described in detail with reference to the accompanying drawings. It should be noted that the embodiments in the present application and the features in the embodiments may be combined with each other without conflict. FIG. 1 is a flowchart of a method for processing voice call quality according to an embodiment of the present invention. As shown in FIG. 1 , the method includes the following steps: Step S102: Set a plurality of scenarios in a mobile terminal, where each scenario corresponds to a set of audio parameters; Step S104: Set the mobile terminal by using audio parameters corresponding to one of the plurality of scenarios. Through the above steps, multiple scenarios are set in the mobile terminal, so that the user is provided with multiple choices, so that the user can select a suitable scenario according to his own needs, to obtain satisfactory voice call quality. The existing mobile terminal cannot judge the user's gender, age level, and mobile terminal use location. In the following preferred embodiment of the present embodiment, according to psychoacoustics and physiological acoustics, for example, users according to different genders or age levels may be used. The spectrum of the auditory and call speech is used to determine the listening ability of different users, and then to set the appropriate scene. For example, at least one of the following determines the audio parameters corresponding to the scene: the age of the user, the gender of the user, and the environment in which the mobile terminal is in a call. In the preferred embodiment, the user can select a separate voice call configuration according to his or her age, gender, and different usage environments in which the voice call is made, which can give the user more voice call quality selection options. The user significantly improves the quality of voice calls without replacing the mobile terminal. In this embodiment, a preferred method for determining an audio parameter corresponding to a scene is also provided (of course, the method for determining an audio parameter may not be used, but the determining method is relatively easy to implement), and the method includes the following Step ^1 Step S1, collecting the audio samples corresponding to the scene; Step S2, using the audio samples to test in the standard anechoic chamber; Step S3, determining the audio parameters corresponding to the scene according to the test results. Preferably, the audio parameters comprise at least one of: parameters for controlling the magnitude of the sound gain in the transmit and/or receive directions, adjusting parameters of the digital gain on the transmit and/or receive channels, adjusting the analog gain on the transmit and/or receive channels The parameters, the parameters of the frequency at which the transmitted and/or received speech are modulated, the parameters that suppress the transmission of the background noise, and the parameters that enhance the effect of the double talk. Of course, in order to be more conveniently implemented in the mobile terminal, the audio parameters corresponding to one of the multiple scenes can be written into a digital signal processing (DSP) register, and the terminal can only use the DSP register. The audio parameters in the settings are made to the mobile terminal. According to the above embodiment and its preferred embodiment, several mobile terminal usage scene modes are preset in the mobile terminal, so that the user can use his or her gender and age level during the call. And the location and environment in which the voice call is made, the scene is selected, and then the mobile terminal dynamically writes the adjusted audio parameters customized according to different scenarios in the DSP of the mobile terminal according to the scene configuration selected by the user. It satisfies the high requirements of users for the quality of voice calls, reduces the after-sales cost of mobile terminal manufacturers, and also makes the voice call quality significantly improved when users use mobile terminals. 2 is a structural block diagram of a processing apparatus for voice call quality according to an embodiment of the present invention. The device is used to implement the foregoing embodiment and its preferred embodiments. The description has been omitted, and the following is related to the structure. The module is explained. As shown in FIG. 2, the device includes: a first setting module 10 and a second setting module 20. The structure will be described below. The first setting module 10 is configured to set a plurality of scenarios in the mobile terminal, where each scenario corresponds to a set of audio parameters; and the second setting module 20 is configured to use audio corresponding to one of the multiple scenarios. The parameters are set for the mobile terminal. 3 is a structural block diagram of a processing apparatus for a preferred voice call quality according to an embodiment of the present invention. As shown in FIG. 3, the apparatus further includes: a parameter preset module 302, configured to determine an audio parameter corresponding to the scene according to at least one of the following: : The age of the user, the gender of the user, and the environment in which the mobile terminal is talking. Preferably, the device further includes: a collection module 304 configured to collect audio samples corresponding to the scene; a test module 306 configured to perform testing in a standard anechoic chamber using the audio samples; and the determining module 308 is configured to The result of the test determines the audio parameters corresponding to the scene. Preferably, the audio parameter corresponding to the scene set by the first setting module 10 includes at least one of the following: a parameter for controlling the size of the sound gain in the transmitting and/or receiving direction, a parameter for adjusting the digital gain on the transmitting and/or receiving channel, Adjust the parameters of the analog gain on the transmit and/or receive channels, the parameters that modulate the frequency of the transmitted and/or received speech, the parameters that suppress the background noise transmission, and the parameters that enhance the double talk effect. Preferably, the device further includes: a writing module 310, configured to write audio parameters corresponding to one of the plurality of scenes into the DSP register, and the terminal sets the mobile terminal according to the audio parameters in the DSP register. The following description is made in conjunction with another preferred embodiment in combination with the above-described embodiments and preferred embodiments thereof. In the preferred embodiment, several types of call usage scenarios are built in the built-in programming software version of the mobile terminal, such as selecting the user's age level, gender, and the real environment in which the user voice calls: family, conference room, road, Beach, bus, etc. Users can set according to their own call needs and actual usage. Through the above three modes, users can choose the voice call quality mode that suits them. In the preferred embodiment, for each scenario, separate different audio parameters are provided in the built-in software of the mobile terminal, so that the user can write the DSP register in real time when the user selects the scene, thereby achieving the purpose of improving the quality of the voice call. In the preferred embodiment, the individual audio parameters of different scenes are obtained by capturing different sound samples, for example, sound samples of different age levels, sound samples of different genders, sound samples of different places of use, and in standard The anechoic chamber passes the test system. For example, the Advanced Communication Quality Analysis (ACQUA) audio test system can be used to test the terminal, adjust the audio parameters in real time, and process the test results, for example, the average subjective score ( The Mean Opinion Score, referred to as the MOS) score size or the ITU-defined voice test standard, determines the audio parameters that use the best results. Then, the MOS score derived from the subjective auditory feeling of the volunteer is used to judge whether the voice call quality at this time is in a good state, and finally whether the selected audio parameter needs to be continuously adjusted. FIG. 4 is a schematic diagram of an environment used by a user body-subject subjective test according to an embodiment of the present invention. As shown in FIG. 4, the voice call quality of the mobile terminal in the sending direction can be tested. If the mobile terminal and the fixed telephone in the two rooms are interchanged, the quality of the voice call in the direction in which the mobile terminal is received can be tested. In addition, for the noise of different use environments, the background noise of the real scene is simulated in the room through the speaker, so as to test the voice call quality of the mobile terminal in different scenarios. In this embodiment, the adjusted audio parameters can also be validated by writing to the DSP register, and the used DSP registers and corresponding algorithms include filters, analog gains, digital gains, echo algorithms, and the like. The following describes the audio-related modules in the mobile terminal. It should be noted that the following modules are merely exemplary, but the implementation of these functions is not limited to being implemented in the following modules. FIG. 5 is a structural block diagram of an audio module of a mobile terminal according to an embodiment of the present invention. As shown in FIG. 5, for mobile terminals, algorithms for generally adjusting voice call quality include Auto Gain Control (AGC) module, digital gain, analog gain, and Finite Impulse Response (FIR). Or no P-monthly response (Infinite Impulse Response, referred to as IIR) Filter and Echo Canceller (EC) modules. Among them, the AGC module is used to control the size of the sound gain in the transmitting and receiving directions, to avoid the excessive or too small sound affecting the user's subjective hearing experience. It is based on the set compression threshold, extended threshold, compression slope, extended slope and static gain registers. Gain adjustment is performed and low frequency noise can be filtered out. Both digital gain and analog gain can increase and decrease the gain on the transmit or receive channels. The FIR or IIR filter is used to modulate the frequency response of the received or transmitted speech, and can be adjusted to achieve optimal conditions according to different scenarios. The EC module is responsible for eliminating echoes during the mobile terminal's call, and the registers of this module can suppress the transmission of background noise and enhance the double talk effect. For different voice call quality modes or scenarios, the above DSP registers need to be adjusted in different scenarios. The following description will be based on the spectrum range in which the sounds are different.
60〜100Hz: 这段频率影响声音的浑厚感, 是低音的基音区。 如果这段频 率很丰满, 音色会显得厚实、 混厚感强。 如果这段频率不足, 音色会变得无 力; 而如果这段频率过强, 音色会出现低频共振声, 有轰呜的感觉。 60~100Hz: This frequency affects the richness of the sound and is the pitch area of the bass. If the frequency is very full, the tone will look thick and thick. If the frequency is insufficient, the tone will become weak; if the frequency is too strong, the tone will have a low frequency resonance and a sensation.
100〜150Hz: 这段频率是影响音色的丰满度, 如果这段频率成分增强, 就会产生一种房间共呜的空间感、 混厚感; 如果这段频率成分缺少, 音色会 变得单薄、 苍白; 如果这段频率成分过强, 音色将会显得浑浊, 语音的清晰 度变差。 150〜300Hz: 这段频率影响声音的力度, 尤其是男声声音的力度。 这段 频率是男声声音的低频度基音频率, 同时也是乐音中和弦的根音频率。 如果 这段频率成分缺乏, 音色会显得发软、 发飘, 语音则会变得软绵绵; 如果这 段频率成分过强, 声音会变得生硬而不自然, 且没有特色。 100~150Hz: This frequency affects the fullness of the tone. If this frequency component is enhanced, it will create a sense of space and thickness in the room. If this frequency component is missing, the tone will become thin. Pale; if this frequency component is too strong, the tone will appear turbid and the clarity of the voice will deteriorate. 150~300Hz: This frequency affects the strength of the sound, especially the strength of the male voice. This frequency is the low-frequency fundamental frequency of the male voice, and is also the root audio frequency of the chord in the tone. If this frequency component is lacking, the tone will appear soft and fluttering, and the voice will become soft. If this frequency component is too strong, the sound will become stiff and unnatural, and there is no special feature.
200~500Hz: 中 氐频段决定声音力度, 如超过 +5dB〜10dB声音变得模糊, 清晰度下降, 下跌 -6〜10dB 声音缺乏力度而显单薄, 音色硬而窄。 200~500Hz: The mid-band frequency determines the sound intensity. If the sound exceeds +5dB~10dB, the sound becomes blurred, the sharpness decreases, and the drop is -6~10dB. The sound lacks strength and is thin, and the sound is hard and narrow.
300〜500Hz: 这段频率是语音的主要音区频率。 这段频率的幅度丰满, 语音有力度。 如果这段频率幅度不足, 声音会显得空洞、 不坚实; 如果这段 频率幅度过强, 音色会变得单调, 相对来说氏频成分少了, 高频成也少了, 语音会变成类似于电话中声音的音色一样, 显得很单调。 500〜lKHz: 频率是人声的基音频率区域, 是一个重要的频率范围。 如 果这段频率丰满, 人声轮廓明朗, 整体感好; 如果这段频率幅度不够, 语音 会产生一种收缩感; 如果这段频率过强,语音就会产生一种向前突出的感觉, 使语音产生一种提前进入人的听觉感受。 800Hz: 这个频率幅度影响音色的力度。 如果这个频率丰满, 音色会显 得强劲有力; 如果这个频率不足, 音色将会显得松驰, 也就是 800Hz以下的 成分特性表现突出了, 低频成分就明显; 而如果这个频率过多, 则会产生喉 音感。 人人都有一个喉腔, 人人都有一定的喉音, 如果音色中的喉音成分过 多了, 则会失掉音色美感。 l~2KHz: 这段频率范围通透感明显, 顺畅感强。 如果这段频率缺乏, 音 色则松散且音色脱节; 如果这段频率过强, 音则有跳跃感。 300~500Hz: This frequency is the main zone frequency of the voice. The frequency of this frequency is full and the voice is strong. If the amplitude of this frequency is insufficient, the sound will appear hollow and not solid; if the frequency is too strong, the tone will become monotonous, the relative frequency component will be less, the high frequency will be less, and the voice will become similar to the phone. The sound of the sound is the same, it looks very monotonous. 500~lKHz: The frequency is the pitch frequency area of the human voice and is an important frequency range. If this frequency is full, the vocal contour is clear and the overall feeling is good; if the frequency is not enough, the voice will have a sense of contraction; if the frequency is too strong, the voice will have a feeling of forward accentuation. The voice produces an auditory feeling that enters the person in advance. 800Hz: This frequency amplitude affects the strength of the tone. If this frequency is full, the tone will appear strong and powerful; if this frequency is insufficient, the tone will appear slack, that is, the characteristic characteristics below 800 Hz will be prominent, and the low frequency component will be obvious; if this frequency is too high, the throat will be produced. Sound sense. Everyone has a throat. Everyone has a certain throat sound. If there are too many throat sounds in the tone, the sound will be lost. l~2KHz: This frequency range has obvious transparency and smoothness. If this frequency is lacking, the tone is loose and the tone is out of line; if this frequency is too strong, the tone has a jump.
2〜3KHz: 这段频率是影响声音明亮度最敏感的频段, 如果这段频率成分 丰富, 则音色的明亮度会增强, 如果这段频率幅度不足, 则音色将会变得朦 朦胧胧;而如果这段频率成分过强,音色就会显得呆板、发硬、不自然。 l〜3KHz 中高频段对明亮度、 清晰度和临场感有重要作用, 此频段超过 +3〜5dB 会使 声音变硬, 超过 +5〜10dB 会出现金属声, 下跌 -3〜5dB 会使声音变硬, 超过 +5〜10dB 会出现金属声,下跌 -3〜5dB 会使音色失去明亮感,下跌 -5〜10dB 声 音发闷, 不清晰。 2~3KHz: This frequency is the most sensitive frequency band that affects the brightness of the sound. If the frequency component is rich, the brightness of the tone will be enhanced. If the frequency is insufficient, the tone will become awkward; If this frequency component is too strong, the tone will appear dull, hard and unnatural. The medium to high frequency band of l~3KHz plays an important role in brightness, sharpness and presence. If the frequency band exceeds +3~5dB, the sound will be hardened. If it exceeds +5~10dB, metal sound will appear, and the drop will be -3~5dB. Hardening, more than +5~10dB will appear metal sound, falling -3~5dB will make the tone lose brightness, falling -5~10dB The sound is boring, not clear.
3〜4KHz: 这个频率的穿透力很强。 人耳耳腔的谐振频率是 1〜4ΚΗζ , 所以人耳对这个频率也是非常敏感的。 如果这段频率成分过少, 听觉能力会 变差, 语音显得模糊不清了。 如果这个频率成分过强, 则会产生咳声的感觉。 针对移动终端持有者的性别, 一般男性声音频谱的低频都比较丰富, 导 致在移动终端的另一侧(即被叫端), 声音听起来会发闷, 这种情况下, 就可 以通过 FIR或 IIR滤波器进行氐频的调整, 抬高氐频部分 100-500HZ之间的 频谱增益, 改善了通话质量。 而女性声音频谱的高频比较丰富, 导致移动终 端的另一侧 (被叫端) 听起来声音发尖和刺耳, 就可以通过 FIR或 IIR滤波 器在高频处 3000-4000HZ之间的频谱增益进行一定的压氏来改善通话质量。 针对移动终端持有者年龄, 人的年龄层次一般分为老年, 中年, 青年, 儿童,根据听觉能力的不同, 声音频谱的分布也不同,就需要使用不同的 FIR 和 IIR滤波器参数进行调节。 如老年人听力比较差, 则需要提高接收通道的 接收强度,如果听筒( Receiver )器件本身支持 t听力辅助( Hear Assist Carrier, 简称为 HAC ) 功能, 则可以打开听筒的 HAC, 但是要保证音量在可以接受 的范围之内, 不能超过人的生理痛阈, 并且不能够超出听筒或扬声器的功率 范围。 针对移动终端持有者的使用环境, 比如马路上, 车流量比较大, 噪声也 比较大,那么就可以在噪声抑制的算法上针对车辆噪声进行更大程度的抑制, 并且还可以通过 AGC对处于马路噪声的频段进行滤波和特殊处理, 以达到 最优的通话效果。 图 6是根据本发明实施例的基于性别, 年龄以及使用场所的场景联系示 意图。 图 6示出了基于性别, 年龄以及使用场所的场景联系图。 基于图 6, 可以提供的使用场景达到 40种左右, 并且根据实际需求还可以进行添加。 基于上述的移动终端使用者的性别, 年龄层次以及使用场所, 需要大约 40 种音频通话场景配置。 每一种场景配置均通过音频实验室釆用不同的音频样 本进行严格测试,模拟实际场景进行测试,保证了每个场景的语音通话质量。 当用户不作任何选择时, 移动终端使用一组默认的音频参数进行配置。 当用 户才艮据自身情况选择了自己的性别, 年龄层次, 以及通话的使用环境, 则移 动终端通过用户的设置进行调用指定场景配置, 进而达到改善语音通话质量 的目的, 大大地提高了移动终端提升语音通话质量的灵活性, 给用户提供了 极大的方便性。 通过上述实施例, 可以满足用户对语音通话质量提出的要求, 降氏了移 动终端厂商售后成本, 也使得用户使用移动终端时, 语音通话质量有了显著 的提升。 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。 以上所述仅为本发明的优选实施例而已, 并不用于限制本发明, 对于本 领域的技术人员来说, 本发明可以有各种更改和变化。 凡在本发明的^"神和 原则之内, 所作的任何修改、 等同替换、 改进等, 均应包含在本发明的保护 范围之内。 3~4KHz: The penetration of this frequency is very strong. The resonant frequency of the human ear cavity is 1 to 4 ΚΗζ, so the human ear is also very sensitive to this frequency. If this frequency component is too small, the hearing ability will be worse, and the voice will be blurred. If this frequency component is too strong, it will produce a coughing sensation. For the gender of the mobile terminal holder, the low frequency of the general male voice spectrum is relatively rich, resulting in the other side of the mobile terminal (ie, the called end), the sound will be boring, in this case, you can pass the FIR Or the IIR filter performs the adjustment of the chirp frequency, and raises the spectral gain between the 100-500HZ portion of the chirp frequency to improve the call quality. The high frequency of the female voice spectrum is rich, which causes the other side (called end) of the mobile terminal to sound sharp and harsh, and can use the FIR or IIR filter to obtain the spectral gain between 3000-4000HZ at high frequencies. Perform a certain pressure to improve the quality of the call. For the age of mobile terminal holders, the age level of people is generally divided into old age, middle age, youth, and children. Depending on the hearing ability, the distribution of sound spectrum is different, and it is necessary to adjust with different FIR and IIR filter parameters. . If the elderly have poor hearing, you need to improve the receiving channel. Receiving strength, if the Receiver device itself supports the Hear Assist Carrier (HAC) function, you can open the HAC of the handset, but ensure that the volume is within acceptable limits and cannot exceed human physiological pain. Threshold and cannot exceed the power range of the earpiece or speaker. For the mobile terminal holder's use environment, such as the road, the traffic volume is relatively large, and the noise is relatively large, then the noise suppression algorithm can be used to suppress the vehicle noise to a greater extent, and can also be in the AGC pair. The frequency band of the road noise is filtered and specially processed to achieve an optimal call effect. 6 is a schematic diagram of scene relationships based on gender, age, and place of use, in accordance with an embodiment of the present invention. Figure 6 shows a scenario linkage diagram based on gender, age, and location of use. Based on Figure 6, there are about 40 usage scenarios that can be provided, and can be added according to actual needs. Based on the gender, age level, and location of the mobile terminal user described above, approximately 40 audio call scene configurations are required. Each scene configuration is rigorously tested by the audio lab using different audio samples, and the actual scene is tested to ensure the voice call quality of each scene. When the user makes no choices, the mobile terminal is configured with a set of default audio parameters. When the user selects his or her gender, age level, and the use environment of the call according to his/her own situation, the mobile terminal invokes the specified scene configuration through the setting of the user, thereby achieving the purpose of improving the quality of the voice call, and greatly improving the mobile terminal. The flexibility to improve the quality of voice calls provides users with great convenience. Through the above embodiments, the requirements of the user for the quality of the voice call can be satisfied, and the cost of the mobile terminal manufacturer is reduced, and the quality of the voice call is significantly improved when the user uses the mobile terminal. Obviously, those skilled in the art should understand that the above 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. The above is only the preferred embodiment of the present invention, and is not intended to limit the present invention, and various modifications and changes can be made to the present invention. Where in the invention ^" God and Within the principles, any modifications, equivalent substitutions, improvements, etc., are intended to be included within the scope of the present invention.

Claims

权 利 要 求 书 Claim
1. 一种语音通话质量的处理方法, 包括: 1. A method for processing voice call quality, comprising:
在移动终端中设置多种场景, 其中, 每一种场景对应一套音频参 数;  Setting a plurality of scenarios in the mobile terminal, where each scenario corresponds to a set of audio parameters;
使用所述多种场景中的一种场景所对应的音频参数对所述移动终 端进行设置。  The mobile terminal is set using audio parameters corresponding to one of the plurality of scenarios.
2. 根据权利要求 1所述的方法, 其中, 根据以下至少之一确定所述场景 所对应的音频参数: 2. The method according to claim 1, wherein the audio parameter corresponding to the scene is determined according to at least one of the following:
用户的年龄段、 用户的性别、 所述移动终端在通话时所处的环境。  The age group of the user, the gender of the user, and the environment in which the mobile terminal is in a call.
3. 居权利要求 1或 2所述的方法, 其中, 釆用以下步骤确定所述场景 对应的音频参数: 3. The method of claim 1 or 2, wherein the following steps are used to determine audio parameters corresponding to the scene:
釆集所述场景对应的音频样本;  Collecting audio samples corresponding to the scene;
使用所述音频样本在标准的消声室进行测试;  Testing the audio sample in a standard anechoic chamber;
才艮据测试的结果确定所述场景对应的音频参数。  The audio parameters corresponding to the scene are determined according to the test results.
4. 根据权利要求 1或 2所述的方法, 其中, 所述音频参数包括以下至少 之一: The method according to claim 1 or 2, wherein the audio parameter comprises at least one of the following:
用于控制发送和 /或接收方向上声音增益大小的参数、 调整发送和 /或接收通道上数字增益的参数、调整发送和 /或接收通道上模拟增益的 参数、 调制发送和 /或接收语音的频率的参数、 抑制背景噪声传输的参 数、 增强双讲通话效果的参数。  Parameters for controlling the magnitude of the sound gain in the transmit and/or receive directions, adjusting the parameters of the digital gain on the transmit and/or receive channels, adjusting the parameters of the analog gain on the transmit and/or receive channels, modulating the transmission and/or receiving speech Frequency parameters, parameters that suppress background noise transmission, and parameters that enhance double talk performance.
5. 根据权利要求 1或 2所述的方法, 其中, 使用所述多种场景中的一种 场景所对应的音频参数对所述移动终端进行设置包括: The method according to claim 1 or 2, wherein setting the mobile terminal by using an audio parameter corresponding to one of the multiple scenarios comprises:
将所述多种场景中的一种场景所对应的音频参数写入到数字信号 处理 DSP寄存器中,所述终端才艮据所述 DSP寄存器中的音频参数对所 述移动终端进行设置。  The audio parameters corresponding to one of the plurality of scenes are written into a digital signal processing DSP register, and the terminal sets the mobile terminal according to the audio parameters in the DSP register.
6. —种语音通话质量的处理装置, 位于移动终端中, 包括: 第一设置模块, 设置为设置多种场景, 其中, 每一种场景对应一 套音频参数; 6. A processing device for voice call quality, located in the mobile terminal, comprising: The first setting module is configured to set a plurality of scenarios, where each scenario corresponds to a set of audio parameters;
第二设置模块, 设置为使用所述多种场景中的一种场景所对应的 音频参数对所述移动终端进行设置。  And a second setting module, configured to set the mobile terminal by using an audio parameter corresponding to one of the multiple scenarios.
7. 根据权利要求 6所述的装置, 其中, 所述装置还包括: The device according to claim 6, wherein the device further comprises:
参数预置模块, 设置为根据以下至少之一确定所述场景所对应的 音频参数:  a parameter preset module, configured to determine an audio parameter corresponding to the scene according to at least one of the following:
用户的年龄段、 用户的性别、 所述移动终端在通话时所处的环境。  The age group of the user, the gender of the user, and the environment in which the mobile terminal is in a call.
8. 根据权利要求 6或 7所述的装置, 其中, 所述装置还包括: The device according to claim 6 or 7, wherein the device further comprises:
釆集模块, 设置为釆集所述场景对应的音频样本;  a collection module, configured to collect audio samples corresponding to the scene;
测试模块, 设置为使用所述音频样本在标准的消声室进行测试; 确定模块 ,设置为才艮据测试的结果确定所述场景对应的音频参数。  The test module is configured to perform testing in the standard anechoic chamber using the audio sample; and the determining module is configured to determine an audio parameter corresponding to the scene according to the test result.
9. 根据权利要求 6或 7所述的装置, 其中, 所述第一设置模块设置的场 景对应的音频参数包括以下至少之一: The device according to claim 6 or 7, wherein the audio parameter corresponding to the scene set by the first setting module comprises at least one of the following:
用于控制发送和 /或接收方向上声音增益大小的参数、 调整发送和 /或接收通道上数字增益的参数、调整发送和 /或接收通道上模拟增益的 参数、 调制发送和 /或接收语音的频率的参数、 抑制背景噪声传输的参 数、 增强双讲通话效果的参数。  Parameters for controlling the magnitude of the sound gain in the transmit and/or receive directions, adjusting the parameters of the digital gain on the transmit and/or receive channels, adjusting the parameters of the analog gain on the transmit and/or receive channels, modulating the transmission and/or receiving speech Frequency parameters, parameters that suppress background noise transmission, and parameters that enhance double talk performance.
10. 根据权利要求 6或 7所述的装置, 其中, 所述装置还包括: The device according to claim 6 or 7, wherein the device further comprises:
写入模块, 设置为将所述多种场景中的一种场景所对应的音频参 数写入到数字信号处理 DSP寄存器中,所述终端根据所述 DSP寄存器 中的音频参数对所述移动终端进行设置。  a writing module, configured to write audio parameters corresponding to one of the plurality of scenarios into a digital signal processing DSP register, where the terminal performs the mobile terminal according to an audio parameter in the DSP register Settings.
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