WO2014161299A1 - 一种语音质量处理的方法及装置 - Google Patents
一种语音质量处理的方法及装置 Download PDFInfo
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- 238000012545 processing Methods 0.000 claims abstract description 143
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L21/00—Speech 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
- G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
- G10L21/0208—Noise filtering
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- G—PHYSICS
- G10—MUSICAL INSTRUMENTS; ACOUSTICS
- G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
- G10L15/00—Speech recognition
- G10L15/01—Assessment or evaluation of speech recognition systems
Definitions
- the present invention relates to the field of communications technologies, and in particular, to a method and apparatus for voice call quality processing in different environments. Background technique
- the purpose of the embodiments of the present invention is to provide a method and a device for processing voice quality, which can better solve the limitation that only one set of voice software algorithm parameters can not meet the user's personalized requirements in different environments. problem.
- An embodiment of the present invention provides a method for processing a voice quality, where the method includes: performing noise detection on a voice signal to be processed to obtain a current noise environment of the voice signal;
- the speech signal is subjected to speech processing using the speech processing parameters such that the speech quality of the processed speech signal is adapted to the current noise environment and the user selected speech quality mode.
- the performing noise detection by the voice signal to be processed, and obtaining a current noise environment of the voice signal includes:
- the digital voice signal is obtained by analog-to-digital conversion of the analog voice signal picked up by the microphone; the current noise environment parameter is obtained by performing noise detection on the digital voice signal; using a preset relationship between the noise environment parameter and the noise environment, The current noise environment (sender) corresponding to the current noise environment parameter.
- the performing noise detection by the voice signal to be processed, and obtaining a current noise environment of the voice signal further includes:
- the current noise environment corresponding to the current noise environment parameter of the other party is obtained.
- the parameter configuration in the found parameter group is selected as the voice processing parameter.
- the performing voice processing on the voice signal by using a voice processing parameter includes: performing noise cancellation processing on the voice signal by using a noise cancellation parameter to cancel noise in the voice signal;
- the speech signal subjected to the denoising process is equalized by the equalization parameter to adjust the distortion of the speech signal.
- the voice processing is performed on the voice signal by using a voice processing parameter, and the method further includes:
- the speech signal is denoised using a noise canceling parameter to cancel noise in the speech signal.
- the voice processing is performed on the voice signal by using a voice processing parameter, and the method further includes:
- the speech signal subjected to the denoising process is subjected to filtering processing using filter parameters to filter out noise outside the frequency band of the speech signal.
- the voice processing is performed on the voice signal by using a voice processing parameter, and the method further includes:
- Denoising the speech signal subjected to the filtering process by using the denoising parameter to cancel the echo in the speech signal;
- the speech signal subjected to the echo cancellation processing is subjected to gain amplification processing to The gain of the speech signal;
- the gain-amplified speech signal is equalized by the equalization parameter to adjust the distortion of the speech signal.
- the embodiment of the present invention further provides a device for processing a voice quality, where the device includes: a noise environment module configured to perform noise detection on a voice signal to be processed to obtain a current noise environment of the voice signal;
- the main processor module is configured to obtain an adapted voice processing parameter according to the current noise environment and a voice quality mode selected by the user;
- the voice processing module is configured to perform voice processing on the voice signal by using the voice processing parameter, so that the voice quality of the processed voice signal is adapted to the current noise environment and the voice quality mode selected by the user.
- the voice processing module includes:
- An equalization submodule configured to perform equalization processing on the voice signal by using an equalization parameter to adjust a distortion of the voice signal
- a filtering submodule configured to filter the speech signal by using a filtering parameter to eliminate noise in the speech signal
- a clearing sub-module configured to perform echo cancellation processing on the speech signal by using a de-sounding parameter to cancel an echo in the speech signal
- the method and device for processing voice quality provided by the embodiments of the present invention can select a required voice quality mode according to different requirements of the user in a specific environment, and obtain the adapted voice quality through different noise environments and different users' voice quality requirements.
- Voice processing parameter method, implementation is not Personalized demand for voice quality in different environments with users.
- FIG. 1 is a flowchart of a method for processing voice quality according to an embodiment of the present invention
- FIG. 4 is a schematic flowchart of a receiver that cooperates between modules of voice quality processing according to an embodiment of the present invention. detailed description
- FIG. 1 is a flowchart of a method for processing voice quality according to an embodiment of the present invention. As shown in FIG. 1, the method includes:
- Step S1 performing noise detection on the speech signal to be processed to obtain a current noise environment of the speech signal.
- Step S2 Obtain an appropriate voice processing parameter according to the current noise environment and the voice quality mode selected by the user.
- Step S3 Perform voice processing on the voice signal by using the voice processing parameter, so that the voice quality of the processed voice signal is adapted to the current noise environment and the voice quality mode selected by the user.
- the current noise environment parameter is obtained; and the relationship between the preset noise environment parameter and the noise environment is used to obtain the current noise environment.
- the current noise environment corresponding to the current noise environment parameter of the other party is obtained.
- step S2 searching for a parameter group that is compatible with the current noise environment and the voice quality mode selected by the user from the saved plurality of parameter groups;
- the parameter configuration in the found parameter group is selected as the voice processing parameter.
- the speech signal subjected to the denoising process is equalized by the equalization parameter to adjust the distortion of the speech signal.
- the speech signal subjected to the denoising process is subjected to filtering processing using filter parameters to filter out noise outside the frequency band of the speech signal.
- the speech signal subjected to the filtering process is subjected to echo cancellation processing by using the echo cancellation parameter to eliminate The echo in the speech signal;
- the gain-amplified speech signal is equalized by the equalization parameter to adjust the distortion of the speech signal.
- Embodiments of the present invention provide a method for implementing a certain voice quality-high fidelity mode in different noise environments.
- 2 is a structural diagram of a device for processing voice quality according to an embodiment of the present invention. As shown in FIG. 2, the method includes: an upper application module 21, a data collection/playback and digital/analog conversion module 22, a voice processing module 23, and parameters. Grouping module 24, main processor module 25.
- the voice processing parameter module in the main processor module is configured to obtain an adapted voice processing parameter according to the current noise environment and a voice quality mode selected by a user.
- the echo sub-module in the voice quality module is configured to perform echo cancellation processing on the voice signal by using a cancellation parameter to cancel echo in the voice signal; and a gain sub-module in the voice quality module, configured The gain amplification process is performed on the speech signal by using a gain parameter to amplify the gain of the speech signal.
- the voice processing module is not limited to a specific implementation, and may use a software processing algorithm or a hardware digital signal processor (DSP) chip.
- the processor module 25 implements the function of the noise environment module, configured to perform noise detection on the speech signal to be processed by the data collection/playback and digital/analog conversion module 22 and the speech processing module 23, and in the main processing The current noise environment of the speech signal is obtained in the module 25.
- the device for processing the voice quality can be set in an electronic device having a voice function, such as a mobile phone, a computer, etc.; the upper application module 21, data collection/playback, and digital/analog in the device.
- it can be a Central Processing Unit (CPU), a Digital Signal Processor (DSP), or a Field-Programmable Gate Array (FPGA) in the device. ) Implementation.
- Step 201 The transmit direction path, the data collection/playback, and the digital-to-analog/analog-to-digital conversion module perform analog-to-digital conversion on the analog voice signal picked up by the microphone. Converted to a digital voice signal.
- Step 202 The digital voice signal is sent to the voice processing module through the digital interface, and the voice processing module first performs noise detection on the digital voice signal to obtain a current noise environment parameter, and sends the data to the main processor module to determine the current user.
- Step 203 After obtaining the parameters of the high fidelity mode required by the user sent by the upper application module, the main processor module uses the high fidelity mode parameter and the current noise environment parameter as related judgment conditions, and queries the parameter grouping module through the relevant interface. Each parameter in the group is grouped.
- Step 204 After obtaining the parameter configuration that satisfies the two conditions from the parameter grouping, reading the parameter configuration through the relevant read interface, and writing the parameter configuration to the voice processing module through the related write interface.
- Step 205 The voice processing module performs corresponding processing on the noise voice signal according to the written parameter configuration, and adjusts the equalization filter and the related high in the high-fidelity mode by using the written spectrum parameter.
- the low-pass filter performs spectral adjustment on the speech signal, and adjusts the frequency band of the distorted speech signal to achieve a distortion-free effect.
- Step 206 Calling the relevant noise cancellation and echo sub-module through the written noise and echo parameters, and performing noise cancellation and echo cancellation processing on the current noise, thereby finally achieving the effect desired by the user.
- Step 207 After the processing is completed, the processed digital voice signal is sent to the main processor module for subsequent processing.
- Step 208 Receive a direction path, and the main processor module sends the decoded digital voice signal to the voice processing module through the digital interface.
- Step 209 The voice processing module performs noise detection on the decoded digital voice signal, obtains a current noise environment parameter, and sends it to the main processor module to determine a noise environment in which the current party is located.
- Step 210 After obtaining the parameters of the high-fidelity mode required by the user sent by the upper-layer application module, the main processor module uses the high-fidelity mode parameter and the current noise environment parameter as related judgment conditions, and queries the grouping module through the relevant interface. Grouping of various parameters.
- Step 211 After obtaining the parameter configuration that satisfies the two conditions from the parameter grouping, the parameter configuration is read through the related read interface. And writing the configuration parameter to the voice processing module through a related write interface.
- Step 212 The speech processing module performs corresponding processing according to the written noise speech signal, and in the high fidelity mode, adjusts the equalization filter and the correlated high and low pass filter through the written spectral parameters, and performs spectrum on the decoded speech signal. Adjustment, by adjusting the frequency band of the distorted speech signal to achieve a distortion-free effect.
- Step 213 Calling the relevant noise canceling and erasing sub-module through the written noise and echo parameters, and performing noise cancellation and echo cancellation processing on the current noise, thereby finally achieving the effect desired by the user.
- Step 214 After the processing is completed, the processed digital voice signal is sent to the data collection/playback and digital-to-analog/analog-to-digital conversion module through the digital interface, and the digital voice signal is converted into an analog voice signal. Send it to the speaker and other devices to play it out.
- the parameter grouping module is grouped based on the current environment of the user and the voice quality mode of the user in the upper application module.
- the environment in which the current user is located is classified according to the noise intensity. : Quiet environment, mild noise environment, moderate noise environment, strong noise environment, etc. It is agreed that the noise intensity of each environment is within a range. For example, in a typical moderate noise environment (such as an office environment), the noise intensity is constant, then the noise intensity in the environment is increased, and when it reaches a certain level, it becomes Strong noise environment. Therefore, when the noise intensity of the user's current environment is within the noise range of a certain noise environment, we will treat it with the typical value of the noise environment.
- the classification of the noise situation is not limited to the specific form described above. It can have other classification forms, such as noise processing in the noise range of each noise environment, and the specific implementation form can be based on the user's needs and specifics. The design is implemented.
- the parameters of high-fidelity mode are divided into multiple parameter groups according to the difference of noise intensity or noise type, and the parameters of high-fidelity mode are adjusted according to different noise environments. Write each parameter grouping.
- the parameters of the high-fidelity mode written in advance are adjusted based on the standard curve of the human voice.
- the effect of the adjustment is different based on the actual spectrum of the speech signal, and thus the parameter configuration is different. .
- the noise intensity is small, and it is not necessary to use a high-low-pass filter to perform strong filtering on the speech signal. Only the noise cancellation sub-module performs a certain intensity of processing to eliminate noise, and adjusts the spectrum by adjusting the spectrum.
- the equalization filter in the sub-module can adjust and repair the distorted speech signal; if the noise environment is relatively bad, in order to achieve the high fidelity of the required speech signal, the noise speech signal cannot be performed by the equalization filter. For strong filtering, the noise speech signal can only be filtered out to some extent by the noise cancellation sub-module. At this time, the high-fidelity signal will be mixed with a slight noise speech signal, that is, the speech signal is exchanged for the noise. Distortion to achieve high fidelity.
- the embodiment of the present invention is only described in the high-fidelity mode in the upper layer application module.
- the main processor reads the parameter configuration in the corresponding noise level in the comfort mode in the parameter grouping module.
- the speech processing module is written, and the speech processing module adjusts and repairs the distorted speech signal in the current noise environment, and filters out the noise through the denoising sub-module, and eliminates the echo in the sub-module by eliminating the sub-module.
- the speech signal is guaranteed to be almost undistorted and almost no noise, so that the speech quality is subjectively comfortable.
- the other steps are similar to the embodiment of the present invention.
- the parameters of the low noise mode are specifically adjusted based on different noise environments, and the parameters of the low noise mode respectively applicable to different noise environments are obtained.
- Configuration For example, when the ambient intensity is a quiet environment, the noise cancellation parameter of the low noise mode that needs to be adjusted is not very strong in processing power; and when the environmental intensity is a strong noise environment, the noise cancellation of the low noise mode needs to be adjusted.
- the parameters make the noise cancellation sub-module have stronger noise processing capability, and the sub-module such as the filter can filter the noise speech signal.
- the partial distortion is sacrificed in exchange for the complete elimination of the unsteady noise, so as to achieve the low noise voice quality mode required by the user. This mode is suitable for users who are harsh in the environment and are not sensitive to the sound distortion. .
- the speech signal needs to be processed specifically.
- the sender after analog gain amplification, the data is collected/played and the digital-to-analog/analog-to-digital conversion module performs analog-to-digital conversion.
- the weaker speech signal is amplified, and the noise signal is amplified.
- the voice signal containing the strong noise voice signal is sent to the voice processing module to obtain the current noise environment parameter, and the main processor module reads the parameter in the private mode according to the current noise environment parameter and the private mode parameter.
- the voice processing module configures the pair in the voice signal according to the read parameter configuration
- the noise signal in the filter is filtered out. Because the noise voice signal is strong, it is necessary to first call the filter sub-module to filter out the noise voice signal outside the voice signal band, and call the noise cancellation sub-module to filter the noise voice signal in the frequency band, and simultaneously call the cancellation sound sub-module to filter out.
- the echo voice signal in the voice signal is finally amplified by digital gain, and the voice signal with less energy is amplified to a suitable gain value, and the spectrum of the voice signal is adjusted by the equalization filter to achieve better fidelity.
- FIG. 3 is a schematic diagram of a sender process of inter-module cooperation between voice quality processing according to an embodiment of the present invention. As shown in FIG. 3, the sender process provided by the present invention is described in detail through specific embodiments, and the steps are as follows:
- Step 301 The user initiates a call request or accepts a called request.
- Step 302 The upper application calls the underlying related process, completes the related initialization work, and invokes the startup hardware process.
- Step 303 The upper application sends the parameter of the voice quality mode selected by the user to the main processing program, and records the parameter in the main processing program.
- Step 304 Start the hardware process, and perform power-on and clock-distribution operations to implement initialization of the software, so that the hardware can work normally.
- Step 305 The microphone sends the picked noise-containing analog voice signal to the data collection/playback and the digital-analog/analog-to-digital conversion module for analog-to-digital conversion.
- Step 306 The data collection/playback and digital/analog conversion module converts the received analog voice signal into a digital voice signal, and then sends the voice signal to the voice processing module through the digital interface for voice processing.
- Step 307 After the voice detection module performs voice detection, the current noise environment parameter is obtained, and the current noise environment parameter is sent to the main processing program through the relevant interface.
- Step 308 The main processing program uses the relevant parameters in step 303 and step 307 as query conditions, performs parameter grouping query in the parameter grouping module, and after querying the parameter configuration that satisfies the condition, reads the relevant parameter configuration, and the read parameter is read. Related parameter configuration is written to the voice processing through the relevant interface. Module.
- Step 309 The voice processing module performs corresponding function processing on the digital voice signal containing the noise echo according to the written related parameter configuration, and sends the function processed digital voice signal to the main processing program again.
- Step 310 The main processing program encodes the digital voice signal processed by the noise echo function, and then sends the encoded digital voice signal to the subsequent processing module for subsequent processing.
- Step 311 The subsequent processing module performs protocol, physical processing, and the like on the encoded digital voice signal, and sends the encoded digital voice signal through the antenna.
- FIG. 4 is a schematic diagram of a receiver software process for mutually cooperating between voice quality processing modules according to an embodiment of the present invention. As shown in FIG. 4, the receiver software flow provided by the present invention is described in detail through specific embodiments, and the steps are as follows:
- Step 401 The user initiates a call request or accepts a called request.
- Step 402 The antenna receives the digital voice signal, and performs corresponding processing, and sends the processed digital voice signal to the main processing program for decoding and the like.
- Step 403 The upper application sends the parameter of the voice quality mode selected by the user to the main processing program, and records the parameter in the main processing program.
- Step 404 The main processing program sends the digital voice signal to the voice processing module for voice detection, and obtains corresponding noise environment parameters.
- Step 405 Send the obtained corresponding noise environment parameter to the main processing program through the relevant interface.
- Step 406 The main processing program performs parameter grouping query in the parameter grouping module according to the relevant parameters in step 404 and step 405 as query conditions. After querying the parameter configuration that satisfies the condition, the related parameter configuration is read, and the related correlation is read. Parameter configuration is written to the voice processing module through the relevant interface.
- Step 407 The voice processing module performs corresponding function processing on the digital voice signal containing the noise echo according to the written related parameter configuration, and sends the function processed digital voice signal to the data collection/playback and digital-to-analog/analog-to-digital conversion. Module.
- Step 408 Perform digital-to-analog conversion of the digital voice signal processed by the function, convert it into an analog voice signal, and play it out.
- Another preferred embodiment of the present invention provides an augmentation method based on an embodiment of the present invention.
- the difference between the expansion method and the embodiment of the present invention is as follows:
- the main processor module controls the speech processor module to close the relevant sub-module without degrading the speech quality, such as turning off automatic gain control (AGC, Automatic Generation). Control ) or Dynamic Range Control (DRC) function.
- AGC automatic gain control
- DRC Dynamic Range Control
- the noise environment parameter of the noise-containing voice signal can be detected once every 2 seconds (s) or longer, so as to reduce the load of the main processor module.
- This expansion solution is not limited to this type of application, but can be combined with other devices to implement other types of functions.
- the condition for configuring the query parameters in the embodiment of the present invention is determined by the voice quality mode selected by the user and the noise environment. On the basis of the embodiment of the present invention, if the user does not select the voice quality mode, the mobile terminal performs query parameter configuration according to the noise environment and the default voice quality mode.
- the specific operation implementation is not limited to the above several cases, depending on the actual needs of the user and the implementation of the design.
- the embodiments of the present invention have the following technical effects: a method for adapting voice processing parameters by using a voice quality mode selected by a user and a current noise environment, thereby realizing different needs of users in a specific environment, and selecting The voice quality mode meets the individual needs of users.
- the embodiments of the present invention can select a voice quality mode according to different requirements of a user in a specific environment, and obtain a suitable voice processing parameter method by using different noise environments and different users' voice quality requirements to implement different users. Personalized demand for voice quality in different environments.
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Abstract
一种语音质量处理的方法及装置,所述方法包括:通过对待处理的语音信号进行噪声检测,得到所述语音信号的当前噪声环境(S1);根据所述当前噪声环境和用户选择的语音质量模式,得到适配的语音处理参数(S2);利用所述语音处理参数对所述语音信号进行语音处理,使处理后的语音信号的语音质量与当前噪声环境和用户选择的语音质量模式相适应(S3)。
Description
一种语音庸量处理的方法及装置 技术领域
本发明涉及通信技术领域, 特别涉及一种不同环境下语音通话质量处 理的方法及装置。 背景技术
当前, 随着智能终端的普及, 许多终端都采用了带有消噪功能的算法 或芯片。 而在当前的终端中, 在手持模式下, 即使用户处于不同的环境, 终端均采用一套语音软件算法参数来对噪声、 回音等进行消除。
随着终端越来越智能、 适用的场景越来越多, 采用一套语音软件算法 参数的局限性就越来越大, 而且不能满足不同用户在不同环境下的主观听 觉要求。 在不同的环境下, 不同身份的人对语音质量也有不同的要求, 在 公共场所或会议室内参加电话会议, 不同的人会有不同的需求, 有人需要 高保真度来区分发言的人, 有的人对噪声大小更敏感, 而有的人对噪声不 敏感, 但对声音的大小要求强烈; 在某些特定的场所或者是特定身份的人, 手机通话双方希望两者谈话如同窃窃私语, 防止被他人听到 ... ..., 这就需 要对硬件的不同模块进行差异性的参数调节来对语音的保真度、 噪声、 回 音等进行相应的处理, 从而满足在不同场景下用户的个性化需求。 发明内容
有鉴于此, 本发明实施例的目的在于提供一种语音质量处理的方法及 装置, 能够更好的解决只能采用一套语音软件算法参数不能满足用户对不 同环境下的个性化需求的局限性问题。
为达到上述目的, 本发明实施例的技术方案是这样实现的:
本发明实施例提供了一种语音质量处理的方法, 所述方法包括: 通过对待处理的语音信号进行噪声检测, 得到所述语音信号的当前噪 声环境;
根据所述当前噪声环境和用户选择的语音质量模式, 得到适配的语音 处理参数;
利用所述语音处理参数对所述语音信号进行语音处理, 使处理后的语 音信号的语音质量与当前噪声环境和用户选择的语音质量模式相适应。
优选地, 所述通过对待处理的语音信号进行噪声检测, 得到所述语音 信号的当前噪声环境包括:
通过对麦克风拾取的模拟语音信号进行模数转换, 得到数字语音信号; 通过对所述数字语音信号进行噪声检测, 得到当前噪声环境参数; 利用预置的噪声环境参数与噪声环境的关系表, 得到与当前噪声环境 参数相对应的 (发送方) 当前噪声环境。
优选地, 所述通过对待处理的语音信号进行噪声检测, 得到所述语音 信号的当前噪声环境还包括:
通过对天线接收的编码的数字语音信号进行解码处理, 得到解码的数 字语音信号;
通过对所述解码的数字语音信号进行噪声检测, 得到当前对方的噪声 环境参数;
利用预置的噪声环境参数与噪声环境的关系表, 得到与当前对方的噪 声环境参数相对应的 (接收方) 当前噪声环境。
优选地, 所述根据所述当前噪声环境和用户选择的语音质量模式, 得 到适配的语音处理参数, 包括:
从保存的多个参数分组中查找与当前噪声环境和用户选择的语音质量 模式相适应的参数分组;
将已查找到的参数分组中的参数配置选作所述语音处理参数。
优选地, 所述利用语音处理参数对所述语音信号进行语音处理, 包括: 利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用均衡参数对进行消噪处理的语音信号进行均衡处理, 以调整所述 语音信号的失真度。
优选地, 所述利用语音处理参数对所述语音信号进行语音处理, 还包 括:
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声。
优选地, 所述利用语音处理参数对所述语音信号进行语音处理, 还包 括:
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用滤波参数对进行消噪处理的语音信号进行滤波处理, 以滤除所述 语音信号中频带以外的噪声。
优选地, 所述利用语音处理参数对所述语音信号进行语音处理, 还包 括:
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用滤波参数对进行消噪处理的语音信号进行滤波处理, 以滤除所述 语音信号中频带以外的噪声;
利用消回音参数对进行滤波处理的语音信号进行消回音处理, 以消除 所述语音信号中的回音;
利用增益参数对进行消回音处理的语音信号进行增益放大处理, 以放
大所述语音信号的增益;
利用均衡参数对进行增益放大的语音信号进行均衡处理, 以调整所述 语音信号的失真度。
本发明实施例还提供了一种语音质量处理的装置, 所述装置包括: 噪声环境模块, 配置为通过对待处理的语音信号进行噪声检测, 得到 所述语音信号的当前噪声环境;
主处理器模块, 配置为根据所述当前噪声环境和用户选择的语音质量 模式, 得到适配的语音处理参数;
语音处理模块, 配置为利用所述语音处理参数对所述语音信号进行语 音处理, 使处理后的语音信号的语音质量与当前噪声环境和用户选择的语 音质量模式相适应。
优选地, 所述语音处理模块包括:
消噪子模块, 配置为利用消噪参数对所述语音信号进行消噪处理, 以 消除所述语音信号中的噪声;
均衡子模块, 配置为利用均衡参数对所述语音信号进行均衡处理, 以 调整所述语音信号的失真度;
滤波子模块, 配置为利用滤波参数对所述语音信号进行滤波处理, 以 消除所述语音信号中的噪声;
消回音子模块, 配置为利用消回音参数对所述语音信号进行消回音处 理, 以消除所述语音信号中的回音;
增益子模块, 配置为利用增益参数对所述语音信号进行增益放大处理, 以放大所述语音信号的增益。
本发明实施例提供的语音质量处理的方法及装置能够根据用户在特定 环境下的不同需求, 选择所需要的语音质量模式, 并通过不同噪声环境以 及不同用户对语音质量的需求, 得到适配的语音处理参数的方法, 实现不
同用户在不同的环境下对语音质量的个性化需求。 附图说明
图 1是本发明实施例提供的语音质量处理的方法流程图;
图 2是本发明实施例提供的语音质量处理的装置结构图;
图 3是本发明实施例提供的语音质量处理的模块间相互协作的发送方 的流程示意图;
图 4是本发明实施例提供的语音质量处理的模块间相互协作的接收方 的流程示意图。 具体实施方式
以下结合附图对本发明的优选实施例进行详细说明, 应当理解, 以下 所说明的优选实施例仅用于说明和解释本发明, 并不用于限定本发明。
图 1是本发明实施例提供的语音质量处理的方法流程图, 如图 1所示, 所述方法包括:
步骤 S1 : 通过对待处理的语音信号进行噪声检测, 得到所述语音信号 的当前噪声环境。
步骤 S2: 根据所述当前噪声环境和用户选择的语音质量模式, 得到适 配的语音处理参数。
步骤 S3 : 利用所述语音处理参数对所述语音信号进行语音处理, 使处 理后的语音信号的语音质量与当前噪声环境和用户选择的语音质量模式相 适应。
所述步骤 S1中, 通过对麦克风拾取的模拟语音信号进行模数转换, 得 到数字语音信号;
通过对所述数字语音信号进行噪声检测, 得到当前噪声环境参数; 利用预置的噪声环境参数与噪声环境的关系表, 得到与当前噪声环境
参数相对应的 (发送方) 当前噪声环境。
通过对天线接收的编码的数字语音信号进行解码处理, 得到解码的数 字语音信号;
通过对所述解码的数字语音信号进行噪声检测, 得到当前对方的噪声 环境参数;
利用预置的噪声环境参数与噪声环境的关系表, 得到与当前对方的噪 声环境参数相对应的 (接收方) 当前噪声环境。
所述步骤 S2中, 从保存的多个参数分组中查找与当前噪声环境和用户 选择的语音质量模式相适应的参数分组;
将已查找到的参数分组中的参数配置选作所述语音处理参数。
所述步骤 S3中, 利用消噪参数对所述语音信号进行消噪处理, 以消除 所述语音信号中的噪声。
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用均衡参数对进行消噪处理的语音信号进行均衡处理, 以调整所述 语音信号的失真度。
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用滤波参数对进行消噪处理的语音信号进行滤波处理, 以滤除所述 语音信号中频带以外的噪声。
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用滤波参数对进行消噪处理的语音信号进行滤波处理, 以滤除所述 语音信号中频带以外的噪声;
利用消回音参数对进行滤波处理的语音信号进行消回音处理, 以消除
所述语音信号中的回音;
利用增益参数对进行消回音处理的语音信号进行增益放大处理, 以放 大所述语音信号的增益;
利用均衡参数对进行增益放大的语音信号进行均衡处理, 以调整所述 语音信号的失真度。
本发明实施例提供了一种解决在不同噪声环境下实现某一语音质量一 一高保真模式的方法。 图 2是本发明实施例提供的语音质量处理的装置结 构图, 如图 2所示, 包括: 上层应用模块 21, 数据收集 /播放及数模 /模数转 换模块 22, 语音处理模块 23, 参数分组模块 24, 主处理器模块 25。
所述主处理器模块中的语音处理参数模块, 配置为根据所述当前噪声 环境和用户选择的语音质量模式, 得到适配的语音处理参数。
所述语音处理模块 23中的语音质量模块, 配置为利用所述语音处理参 数对所述语音信号进行语音处理, 使处理后的语音信号的语音质量与当前 噪声环境和用户选择的语音质量模式相适应。 其中, 所述语音质量模块中 的消噪子模块, 配置为利用消噪参数对所述语音信号进行消噪处理, 以消 除所述语音信号中的噪声; 所述语音质量模块中的均衡子模块, 配置为利 用均衡参数对所述语音信号进行均衡处理, 以调整所述语音信号的失真度; 所述语音质量模块中的滤波子模块, 配置为利用滤波参数对所述语音信号 进行滤波处理, 以消除所述语音信号中的噪声。 所述语音质量模块中的消 回音子模块, 配置为利用消回音参数对所述语音信号进行消回音处理, 以 消除所述语音信号中的回音; 所述语音质量模块中的增益子模块, 配置为 利用增益参数对所述语音信号进行增益放大处理, 以放大所述语音信号的 增益。 其中语音处理模块不局限于具体的实现, 可以使用软件的处理算法, 也可以使用硬件的数字信号处理器(DSP, Digital Signal Processor ) 芯片。
所述数据收集 /播放及数模 /模数转换模块 22、 语音处理模块 23和主处
理器模块 25实现了噪声环境模块的功能,配置为通过所述数据收集 /播放及 数模 /模数转换模块 22和语音处理模块 23对待处理的语音信号进行噪声检 测, 并在所述主处理器模块 25中得到所述语音信号的当前噪声环境。
其中, 所述语音质量处理的装置在实际应用中, 可设置于具有语音功 能的电子设备中, 例如手机、 电脑等; 所述装置中的上层应用模块 21、 数 据收集 /播放及数模 /模数转换模块 22、 语音处理模块 23及其子模块: 语音 质量模块、 消噪子模块、 均衡子模块、 滤波子模块、 消回音子模块、 增益 子模块、 参数分组模块 24和主处理器模块 25, 在实际应用中, 均可由装置 中的中央处理器 (Central Processing Unit, CPU ), 数字信号处理器(DSP, Digital Signal Processor )、 或可编程门阵歹 'J ( Field-Programmable Gate Array, FPGA ) 实现。
所述装置的工作原理如下, 分别对发送方和接收方进行说明: 步骤 201 : 发送方向通路, 数据收集 /播放及数模 /模数转换模块将麦克 风拾取到的模拟语音信号进行模数转换, 转换成数字语音信号。
步骤 202: 将数字语音信号通过数字接口送入语音处理模块,语音处理 模块首先对所述数字语音信号进行噪声检测, 获得当前的噪声环境参数, 并将其送入主处理器模块来确定当前用户所处的噪声环境。
步骤 203:主处理器模块在获得上层应用模块下发的用户需要的高保真 模式的参数后, 将所述高保真模式参数和当前的噪声环境参数作为相关判 断条件, 通过相关接口查询参数分组模块中的各个参数分组。
步骤 204: 从参数分组中得到满足两者条件的参数配置后, 通过相关读 接口, 读取所述参数配置, 并通过相关写接口将所述参数配置写入语音处 理模块。
步骤 205:语音处理模块按照写入的参数配置对噪声语音信号进行相应 处理, 在高保真模式下, 通过写入的频谱参数, 调节均衡滤波器和相关高
低通滤波器, 对语音信号进行频谱调节, 通过对失真的语音信号频段进行 调节, 使其达到无失真的效果。
步骤 206: 通过写入的噪声、 回音参数,调用相关消噪、 消回音子模块, 对当前的噪声进行消噪消回音处理, 最终达到用户需要的效果。
步骤 207: 处理完成后, 将处理后的数字语音信号送入主处理器模块进 行后续处理。
步骤 208: 接收方向通路, 主处理器模块将解码后的数字语音信号通过 数字接口送入语音处理模块。
步骤 209: 语音处理模块对解码后的数字语音信号进行噪声检测, 获得 当前的噪声环境参数, 并将其送入主处理器模块来确定当前对方所处的噪 声环境。
步骤 210:主处理器模块在获得上层应用模块下发的用户需要的高保真 模式的参数后, 将所述高保真模式参数和当前的噪声环境参数作为相关判 断条件, 通过相关接口查询分组模块中的各个参数分组。
步骤 211 : 从参数分组中得到满足两者条件的参数配置后, 通过相关读 接口, 读取所述参数配置。 并通过相关写接口将所述配置参数写入语音处 理模块.
步骤 212: 语音处理模块按照写入的对噪声语音信号进行相应处理, 在 高保真模式下, 通过写入的频谱参数, 调节均衡滤波器和相关高低通滤波 器, 对解码后的语音信号进行频谱调节, 通过对失真的语音信号频段进行 调节, 使其达到无失真的效果。
步骤 213: 通过写入的噪声、 回音参数,调用相关消噪、 消回音子模块, 对当前的噪声进行消噪消回音处理, 最终达到用户需要的效果。
步骤 214: 处理完成后, 将处理后的数字语音信号通过数字接口送入数 据收集 /播放及数模 /模数转换模块, 将数字语音信号转换成模拟语音信号,
送入扬声器等器件播放出来即可。
在上述的基本原理中, 参数分组模块基于用户当前所处的环境以及上 层应用模块中用户需求的语音质量模式进行分组, 那么在各实施例中, 将 当前用户所处的环境按照噪声强度分为: 安静环境、 轻微噪声环境、 中度 噪声环境、 强噪声环境等。 其中约定每一种环境的噪声强度处于一个范围 内, 例如典型的中度噪声环境(如办公环境) 内, 噪声强度是一定的, 那 么增大此环境内的噪声强度, 当达到一定程度后成为强噪声环境。 所以当 用户当前所处环境的噪声强度处于某一个噪声环境的噪声范围内时, 我们 就以此噪声环境的典型值进行处理。 当然对于噪声情况的分类不拘泥于具 体的上述形式, 它可以有其它的分类形式, 如每种噪声环境设置噪声范围 内的最大值进行噪声处理等, 具体的实现形式可以依据用户的需求及具体 的设计来实现。
在高保真模式下, 基于噪声环境的不同, 高保真模式的参数按照噪声 强度的不同或者噪声类型的不同分为多个参数分组, 并将高保真模式的参 数按照不同的噪声环境进行调节, 重新写入各个参数分组。 其中事先写入 的高保真模式的参数是基于人的声音的标准曲线进行调节的, 在不同的噪 声环境下, 基于实际的语音信号频谱, 调节的效果会不同, 从而得到的参 数配置也不一样。 例如在较安静环境下, 噪声强度较小, 不需要使用高低 通滤波器对语音信号进行较强的滤除, 只需要消噪子模块进行一定强度的 处理就能消除噪声, 并通过调节频谱调节子模块中的均衡滤波器来对失真 的语音信号进行调节修复即可; 如果所处的噪声环境比较恶劣, 为了使需 要的语音信号达到高保真的程度, 不能通过均衡滤波器对噪声语音信号进 行较强的滤除, 只能通过消噪子模块对噪声语音信号进行一定程度的滤除, 此时高保真的信号中会混有轻微的噪声语音信号, 即以噪声为代价换取语 音信号的不失真, 从而达到高保真的效果。
本发明实施例只是以上层应用模块中的高保真模式来进行说明, 在通 话的过程中, 如果用户选用舒适度模式, 步骤同本发明实施例类似, 即在 不同的噪声环境下, 主处理器模块会读取参数分组模块中舒适度模式下相 应噪声强度中的参数配置。 通过此参数配置, 写入语音处理模块, 语音处 理模块会对当前噪声环境下的失真语音信号进行调节修复, 并通过消噪子 模块滤除其中的噪声, 通过消回音子模块消除其中的回音。 最后对语音信 号进行整体的处理后, 保证语音信号几乎不失真与几乎没有噪声, 使得语 音音质达到主观比较舒适的效果。
如果用户选用低噪声模式, 其他步骤同本发明实施例类似, 不同点在 于基于不同的噪声环境, 对低噪声模式的参数会进行特定的调节, 得到分 别适用于不同噪声环境的低噪声模式的参数配置。 例如当环境强度为安静 环境时, 对于需要调节的低噪声模式的消噪参数, 使其达到处理能力不是 很强的效果; 而当环境强度为强噪声环境时, 需要调节低噪声模式的消噪 参数, 使得消噪子模块的噪声处理能力比较强, 滤波器等子模块能够对噪 声语音信号进行滤除。 在此种情况下, 以牺牲部分失真度来换取对非稳态 噪声的全部消除, 从而达到用户需要的低噪声语音质量模式, 此模式适用 于环境较恶劣, 同时对声音失真度不敏感的用户。
如果用户选用私密模式, 因为在此模式下, 输入的不含噪声的语音信 号能量较小, 噪声语音信号较强, 故需要对语音信号进行特定的处理。 对 发送方来说, 通过模拟增益放大后, 经过数据收集 /播放及数模 /模数转换模 块进行模数转换。 其中经过模拟增益放大后, 较弱的语音信号被放大, 同 时放大的还有噪声语音信号。 此时将含有较强噪声语音信号的语音信号送 入语音处理模块, 得到当前的噪声环境参数, 主处理器模块根据当前的噪 声环境参数和私密模式的参数, 读取参数分组模块中私密模式下相应噪声 强度中的参数配置, 语音处理模块按照读取的参数配置对夹杂在语音信号
中的噪声语音信号进行滤除。 因为噪声语音信号较强, 所以需要先调用滤 波器子模块滤除语音信号频带外的噪声语音信号, 调用消噪子模块对频带 内的噪声语音信号进行滤除, 同时调用消回音子模块滤除语音信号中的回 音语音信号, 最后经过数字增益放大, 将能量较小的语音信号放大到合适 的增益值, 经过均衡滤波器, 调节语音信号的频谱, 使其达到较好的保真 度。
图 3是本发明实施例提供的语音质量处理的模块间相互协作的发送方 流程示意图。 如图 3 所示, 通过具体实施例对本发明提供的发送方流程进 行详细描述, 步骤如下:
步骤 301 : 用户发起呼叫请求或接受被叫请求。
步骤 302: 上层应用程序调用底层的相关流程, 完成相关初始化工作, 并调用启动硬件流程。
步骤 303 :上层应用程序会把用户选择的语音质量模式的参数发送给主 处理程序, 并将所述参数记录在主处理程序中。
步骤 304: 启动硬件流程, 通过对硬件进行上电、 分配时钟等操作, 实 现对软件的初始化操作, 使得硬件能够正常工作。
步骤 305 : 麦克风将拾取的含有噪声的模拟语音信号发送给数据收集 / 播放及数模 /模数转换模块进行模数转换。
步骤 306: 数据收集 /播放及数模 /模数转换模块对收到的模拟语音信号 转换成数字语音信号后, 通过数字接口发送给语音处理模块进行语音处理。
步骤 307:经语音处理模块进行语音检测后,得到当前的噪声环境参数, 并将当前的噪声环境参数通过相关接口送入主处理程序。
步骤 308:主处理程序将步骤 303和步骤 307中的相关参数作为查询条 件, 在参数分组模块中进行参数分组查询, 查询到满足条件的参数配置后, 读取相关参数配置, 将读取到的相关参数配置通过相关接口写入语音处理
模块。
步骤 309: 语音处理模块按照写入的相关参数配置,对含有噪声回音的 数字语音信号进行相应的功能处理, 并将功能处理后的数字语音信号再次 发送给主处理程序。
步骤 310:主处理程序对经过噪声回音功能处理后的数字语音信号进行 编码等工作后, 将编码后的数字语音信号发送给后续处理模块进行后续处 理。
步骤 311 : 后续处理模块对编码后的数字语音信号进行协议、 物理等处 理, 通过天线将编码后的数字语音信号发送出去。
图 4是本发明实施例提供的语音质量处理的模块间相互协作的接收方 软件流程示意图。 如图 4所示, 通过具体实施例对本发明提供的接收方软 件流程进行详细描述, 步骤如下:
步骤 401 : 用户发起呼叫请求或接受被叫请求。
步骤 402: 天线将数字语音信号接收下来, 并进行相应的处理, 将处理 完的数字语音信号送入主处理程序, 进行解码等工作。
步骤 403:上层应用程序将用户选择的语音质量模式的参数发送给主处 理程序, 并将所述参数记录在主处理程序中。
步骤 404: 主处理程序将数字语音信号送入语音处理模块进行语音检 测, 得到相应的噪声环境参数。
步骤 405:将得到的相应的噪声环境参数通过相关接口发送给主处理程 序。
步骤 406:主处理程序按照步骤 404和步骤 405中的相关参数作为查询 条件在参数分组模块中进行参数分组查询, 查询到满足条件的参数配置后, 读取相关参数配置, 将读取到的相关参数配置通过相关接口写入语音处理 模块。
步骤 407: 语音处理模块按照写入的相关参数配置,对含有噪声回音的 数字语音信号进行相应的功能处理, 并将功能处理后的数字语音信号发送 给数据收集 /播放及数模 /模数转换模块。
步骤 408: 将功能处理后的数字语音信号进行数模转换, 转换成模拟语 音信号, 播放出来即可。
本发明另一优选实施例提供了一种在本发明实施例基础上的扩充方 法, 所述扩充方法相比较于本发明实施例的区别如下:
配合移动终端中的加速传感器等器件, 可以检测到当外部环境安静时, 且用户处于非运动状态。 在此种情况下, 通过调节相关语音质量模式的参 数, 在不损害语音质量的前提下, 由主处理器模块控制语音处理器模块来 关闭相关子模块,如关闭自动增益控制( AGC, Automatic Generation Control ) 或动态范围控制(DRC, Dynamic Range Control )功能等。 这样, 在长时间 通话的过程中, 可以减小相关子模块的负载, 减小发热耗电问题; 也可以 在此种情况下, 不必每次都对含有噪声的语音信号进行查询参数配置并读 取参数配置的工作, 可以通过每 2秒(s )或更长时间对含有噪声的语音信 号的噪声环境参数进行一次检测, 达到减小主处理器模块负载的目的。 此 扩充方案也不仅仅局限于此一类应用, 还可以配合其他器件, 实现其他一 类的功能。
本发明又一优选实施例提供了一种在本发明实施例基础上的扩充方 法, 所述扩充方法相比较于本发明实施例的区别如下:
本发明实施例进行查询参数配置的条件是由用户选择的语音质量模式 与噪声环境共同决定的。 在本发明实施例的基础上, 如果用户没有进行语 音质量模式的选择, 移动终端会根据噪声环境及默认的语音质量模式进行 查询参数配置。 具体的操作实现不局限于上述几种情况, 取决于用户的实 际需求及设计的实现。
综上所述, 本发明实施例具有以下技术效果: 通过用户选择的语音质 量模式与当前的噪声环境得到适配的语音处理参数的方法, 实现用户在特 定环境下的不同需求, 通过选择所需要的语音质量模式, 满足用户的个性 化需求。
尽管上文对本发明实施例进行了详细说明, 但是本发明实施例不限于 按照本发明实施例原理所作的修改, 都应当理解为落入本发明实施例的保 护范围。 工业实用性
本发明实施例能够根据用户在特定环境下的不同需求, 选择所需要的 语音质量模式, 并通过不同噪声环境以及不同用户对语音质量的需求, 得 到适配的语音处理参数的方法, 实现不同用户在不同的环境下对语音质量 的个性化需求。
Claims
1、 一种语音质量处理的方法, 所述方法包括:
通过对待处理的语音信号进行噪声检测, 得到所述语音信号的当前噪 声环境;
根据所述当前噪声环境和用户选择的语音质量模式, 得到适配的语音 处理参数;
利用所述语音处理参数对所述语音信号进行语音处理, 使处理后的语 音信号的语音质量与当前噪声环境和用户选择的语音质量模式相适应。
2、 根据权利要求 1所述的方法, 其中, 所述通过对待处理的语音信号 进行噪声检测, 得到所述语音信号的当前噪声环境包括:
通过对麦克风拾取的模拟语音信号进行模数转换, 得到数字语音信号; 通过对所述数字语音信号进行噪声检测, 得到当前噪声环境参数; 利用预置的噪声环境参数与噪声环境的关系表, 得到与当前噪声环境 参数相对应的当前噪声环境。
3、 根据权利要求 1或 2所述的方法, 其中, 所述通过对待处理的语音 信号进行噪声检测, 得到所述语音信号的当前噪声环境还包括:
通过对天线接收的编码的数字语音信号进行解码处理, 得到解码的数 字语音信号;
通过对所述解码的数字语音信号进行噪声检测, 得到当前对方的噪声 环境参数;
利用预置的噪声环境参数与噪声环境的关系表, 得到与当前对方的噪 声环境参数相对应的当前噪声环境。
4、 根据权利要求 1所述的方法, 其中, 所述根据所述当前噪声环境和 用户选择的语音质量模式, 得到适配的语音处理参数, 包括:
从保存的多个参数分组中查找与当前噪声环境和用户选择的语音质量
模式相适应的参数分组;
将已查找到的参数分组中的参数配置选作所述语音处理参数。
5、 根据权利要求 1所述的方法, 其中, 所述利用语音处理参数对所述 语音信号进行语音处理, 包括:
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声。
6、 根据权利要求 1所述的方法, 其中, 所述利用语音处理参数对所述 语音信号进行语音处理, 包括:
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用均衡参数对进行消噪处理的语音信号进行均衡处理, 以调整所述 语音信号的失真度。
7、 根据权利要求 1所述的方法, 其中, 所述利用语音处理参数对所述 语音信号进行语音处理, 包括:
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用滤波参数对进行消噪处理的语音信号进行滤波处理, 以滤除所述 语音信号中频带以外的噪声。
8、 根据权利要求 1所述的方法, 其中, 所述利用语音处理参数对所述 语音信号进行语音处理, 包括:
利用消噪参数对所述语音信号进行消噪处理, 以消除所述语音信号中 的噪声;
利用滤波参数对进行消噪处理的语音信号进行滤波处理, 以滤除所述 语音信号中频带以外的噪声;
利用消回音参数对进行滤波处理的语音信号进行消回音处理, 以消除
所述语音信号中的回音;
利用增益参数对进行消回音处理的语音信号进行增益放大处理, 以放 大所述语音信号的增益;
利用均衡参数对进行增益放大的语音信号进行均衡处理, 以调整所述 语音信号的失真度。
9、 一种语音质量处理的装置, 所述装置包括:
噪声环境模块, 配置为通过对待处理的语音信号进行噪声检测, 得到 所述语音信号的当前噪声环境;
语音处理参数模块, 配置为根据所述当前噪声环境和用户选择的语音 质量模式, 得到适配的语音处理参数;
语音质量模块, 配置为利用所述语音处理参数对所述语音信号进行语 音处理, 使处理后的语音信号的语音质量与当前噪声环境和用户选择的语 音质量模式相适应。
10、 根据权利要求 9所述的装置, 其中, 所述语音质量模块包括: 消噪子模块, 配置为利用消噪参数对所述语音信号进行消噪处理, 以 消除所述语音信号中的噪声;
均衡子模块, 配置为利用均衡参数对所述语音信号进行均衡处理, 以 调整所述语音信号的失真度;
滤波子模块, 配置为利用滤波参数对所述语音信号进行滤波处理, 以 消除所述语音信号中的噪声;
消回音子模块, 配置为利用消回音参数对所述语音信号进行消回音处 理, 以消除所述语音信号中的回音;
增益子模块, 配置为利用增益参数对所述语音信号进行增益放大处理, 以放大所述语音信号的增益。
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CN113531844A (zh) * | 2021-07-06 | 2021-10-22 | 青岛海尔空调器有限总公司 | 空调器降噪的控制方法、系统、电子设备和存储介质 |
CN113808605A (zh) * | 2021-09-29 | 2021-12-17 | 睿云联(厦门)网络通讯技术有限公司 | 一种基于楼宇对讲系统的语音增强方法和装置以及设备 |
CN113808605B (zh) * | 2021-09-29 | 2023-09-12 | 睿云联(厦门)网络通讯技术有限公司 | 一种基于楼宇对讲系统的语音增强方法和装置以及设备 |
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