WO2020024807A1 - Artificial cochlea ambient sound sensing method and system - Google Patents

Artificial cochlea ambient sound sensing method and system Download PDF

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WO2020024807A1
WO2020024807A1 PCT/CN2019/096648 CN2019096648W WO2020024807A1 WO 2020024807 A1 WO2020024807 A1 WO 2020024807A1 CN 2019096648 W CN2019096648 W CN 2019096648W WO 2020024807 A1 WO2020024807 A1 WO 2020024807A1
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module
sound
neural network
feature values
classification
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French (fr)
Chinese (zh)
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张晓薇
韩彦
孙晓安
黄穗
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浙江诺尔康神经电子科技股份有限公司
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Priority to ES202190003A priority Critical patent/ES2849124B2/en
Publication of WO2020024807A1 publication Critical patent/WO2020024807A1/en

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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception
    • H04R25/50Customised settings for obtaining desired overall acoustical characteristics
    • H04R25/505Customised settings for obtaining desired overall acoustical characteristics using digital signal processing
    • H04R25/507Customised settings for obtaining desired overall acoustical characteristics using digital signal processing implemented by neural network or fuzzy logic
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L15/00Speech recognition
    • G10L15/02Feature extraction for speech recognition; Selection of recognition unit
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L15/00Speech recognition
    • G10L15/06Creation of reference templates; Training of speech recognition systems, e.g. adaptation to the characteristics of the speaker's voice
    • G10L15/063Training
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L15/00Speech recognition
    • G10L15/08Speech classification or search
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00Speech or voice signal processing techniques to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208Noise filtering
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/27Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the analysis technique
    • G10L25/30Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 characterised by the analysis technique using neural networks
    • GPHYSICS
    • G10MUSICAL INSTRUMENTS; ACOUSTICS
    • G10LSPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00
    • G10L25/48Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use
    • G10L25/51Speech or voice analysis techniques not restricted to a single one of groups G10L15/00 - G10L21/00 specially adapted for particular use for comparison or discrimination
    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R25/00Deaf-aid sets, i.e. electro-acoustic or electro-mechanical hearing aids; Electric tinnitus maskers providing an auditory perception

Definitions

  • the invention belongs to the field of signal processing, and relates to a method and system for sensing ambient sound of an artificial cochlea.
  • Cochlear implants are currently the only medical device on the market that can effectively restore hearing in patients with severe or severe deafness.
  • the general working principle of a cochlear implant is to convert the sound signal collected by the microphone into a stimulus code and send it to the implant through the signal processing unit. The implant then stimulates the auditory nerve through microelectrodes according to the stimulus code, so that the implant Restore hearing.
  • this type of system lacks an important function of ordinary human hearing systems, which is to distinguish target signals in complex sound scenes and extract them. For example, in a group of people or in a relatively noisy environment, listen to what the chat object is saying. The usual solution is to reduce the effect of noise on the sound of hearing through a certain denoising algorithm.
  • the denoising algorithm and the parameter configuration of the algorithm are different in different environments (such as pure speech, speech to be noisy, or noise environment).
  • the ambient sound perception algorithm is also introduced. According to the judgment result of the ambient sound perception algorithm, the system can open the noise reduction algorithm and configure related parameters in a targeted manner.
  • the classifier of the ambient sound perception algorithm used a hidden Markov model. The model is relatively simple, the theory matures earlier, and it does not require high training data. It also maintains a certain correct recognition rate. And its calculation volume is relatively low, and it can adapt to such a device with limited computing power as a cochlear implant. With the continuous innovation of pattern recognition, machine learning and other fields in recent years, and the continuous improvement of computing power algorithms, more classification algorithms (support vector machines, neural networks, etc.) have become more prominent in the field of ambient sound perception.
  • classifiers such as support vector machines and neural networks focus on distinguishing categories compared to hidden Markov models without providing prior probabilities of class conversion. In other words, only the data of different ambient sounds need to be analyzed, and the probability of converting one ambient sound to another ambient sound need not be considered. Obtaining this conversion probability is very difficult, and the analysis from the data is not accurate enough.
  • neural networks have many changes. According to the number of input feature values, the number of hidden layers, and the number of network nodes in each layer, the network structure can have many combinations. Moreover, the classification accuracy rate of neural networks is usually proportional to its scale, so the amount of computation required is also relatively large.
  • the present invention proposes an environmental sound sensing method of an artificial cochlea in response to the shortcomings of the existing sound perception processing.
  • the neural sound network is used to classify the environmental sound.
  • the optimization on the cochlear system is to minimize the amount of calculations when a certain classification accuracy is met.
  • the technical solution of the present invention is a method for sensing an ambient sound of a cochlear implant, which includes the following steps:
  • the sound collection module uses a microphone to collect the ambient sound in real time, and then outputs the collected discrete sound signal to the sound feature extraction module;
  • the sound feature extraction module processes the sound signals sent by the sound acquisition module, extracts a set of feature values representing the characteristics of the sound signals, and outputs them to the neural network classification module;
  • the neural network classification module After receiving a set of feature values extracted by the sound feature extraction module, the neural network classification module classifies the set of feature values through a trained neural network, and then outputs the classification result to the comprehensive decision module;
  • the comprehensive decision module After receiving the classification result of the neural network classification module, the comprehensive decision module comprehensively analyzes and gives the judgment of the current scene, and outputs the judgment result to the speech processing selection module;
  • the speech processing selection module selects the optimal speech processing program and its parameter configuration according to the judgment result of the current scene by the comprehensive decision module.
  • the microphone collects ambient sounds in real time using an omnidirectional microphone or a microphone array.
  • the sampling rate of the sound acquisition module is 16K.
  • the feature value extracted by extracting a set of feature values representing the characteristics of the sound signal is eight.
  • the neural network classification module uses a deep neural network or a delayed neural network including two hidden layers and 15 neurons in each layer.
  • the eight characteristic values are selected from 60 characteristic values.
  • the feature value screening adopts a method of comprehensively analyzing the statistical value of the feature value and the Gaussian mixture model, the average influence value algorithm, the sequence forward selection algorithm, and the evaluation of the classifier training result.
  • the calculation amount of the characteristic value and the calculation amount of the neural network does not exceed 20% of the calculation capacity of the cochlear speech processor.
  • the present invention also provides an environmental sound perception system for a cochlear implant, which includes a sound acquisition module, a sound feature extraction module, a neural network classification module, a comprehensive decision module, and a speech processing selection module, which are connected in this order.
  • the sound collection module is configured to use a microphone to collect ambient sounds in real time, and then output the collected discrete sound signal to a sound feature extraction module;
  • the sound feature extraction module is configured to process the sound signals sent by the sound acquisition module, extract a set of feature values representing the characteristics of the sound signal, and output to a neural network classification module;
  • the neural network classification module is configured to, after receiving a set of feature values extracted by the sound feature extraction module, classify the set of feature values through a trained neural network, and then output the classification result to a comprehensive decision module;
  • the comprehensive decision module is configured to comprehensively analyze and give a judgment of the current scene after receiving the classification result of the neural network classification module, and output the judgment result to the voice processing selection module;
  • the speech processing selection module is configured to select an optimal speech processing program and its parameter configuration according to the determination result of the current scenario by the comprehensive decision module.
  • FIG. 1 is a flowchart of steps in a method for sensing an environmental sound of a cochlear implant according to an embodiment of the present invention
  • FIG. 2 is a structural block diagram of an environmental sound sensing system of a cochlear implant according to an embodiment of the present invention
  • FIG. 3 is a specific schematic diagram of a neural network classification module of a method and a system for sensing an environmental sound of a cochlear implant according to an embodiment of the present invention
  • FIG. 4 is a comparison diagram of a calculation amount and an accuracy rate of a method of sensing an ambient sound of a cochlear implant on networks of different hidden layers and different numbers of neurons according to an embodiment of the present invention.
  • FIG. 1 a flowchart of steps of a method for sensing an environmental sound of a cochlear implant according to an embodiment of the present invention is provided, including the following steps:
  • the sound collection module uses a microphone to collect ambient sounds in real time, and then outputs the collected discrete sound signal to the sound feature extraction module;
  • the sound feature extraction module processes the sound signals sent by the sound acquisition module, extracts a set of feature values representing the characteristics of the sound signal, and outputs them to the neural network classification module;
  • the neural network classification module classifies the set of feature values through a trained neural network, and then outputs the classification result to the comprehensive decision module;
  • the comprehensive decision module After receiving the classification result of the neural network classification module, the comprehensive decision module comprehensively analyzes and gives a judgment of the current scene, and outputs the judgment result to the voice processing selection module;
  • the voice processing selection module selects an optimal voice processing program and its parameter configuration according to the judgment result of the current scenario by the comprehensive decision module.
  • FIG. 2 An embodiment of the system of the present invention is shown in FIG. 2 and includes a sound collection module 10, a sound feature extraction module 20, a neural network classification module 30, a comprehensive decision module 40, and a voice processing selection module 50, which are connected in this order.
  • a sound collection module 10 configured to use a microphone to collect ambient sounds in real time, and then output the collected discrete sound signal to the sound feature extraction module 20;
  • the sound feature extraction module 20 is configured to process the sound signal sent by the sound acquisition module, extract a set of feature values representing the characteristics of the sound signal, and output it to the neural network classification module 30;
  • a neural network classification module 30 is configured to classify the set of feature values through a trained neural network after receiving a set of feature values extracted by the sound feature extraction module, and then output the classification result to the comprehensive decision module 40;
  • the comprehensive decision module 40 is configured to, after receiving the classification result of the neural network classification module, comprehensively give a determination of the current scene, and output the determination result to the speech processing selection module 50;
  • the voice processing selection module 50 is configured to select an optimal voice processing program and its parameter configuration according to the judgment result of the current scene by the comprehensive decision module.
  • an omnidirectional microphone or a microphone array is used by the microphone to collect ambient sounds in real time in S10, and the sampling rate of the sound collection module 10 is 16K.
  • a set of feature values representing the characteristics of the sound signal is extracted in S20.
  • the extracted feature values are eight, and the eight feature values are filtered from the 60 feature values.
  • the normalization process is performed before the feature values are extracted. The formula is as follows:
  • x norm is the normalized result
  • X max is the maximum value of the training sample where the feature value is
  • X min is the minimum value of the training sample where the feature value is.
  • the neural network classification module in S30 uses a deep neural network or a delayed neural network with two hidden layers and 15 neurons in each layer.
  • the neural network module is obtained through training on a large number of data samples. Taking the discrimination of 4 types of environmental sounds (pure speech, noisy speech, noise, music, and quietness) as examples, see Figure 3 for its neural network model.
  • the eigenvalues are selected from 1, 2, 3, 4, 5, and 6, and a total of six types form a group.
  • the training samples are extracted from a large number of collected audio files, and they contain a total of 144,000 sets of sample feature values, and each type of ambient sound contains 36,000 sets of feature values.
  • each layer has a different number of neurons. It can be seen from the figure that the accuracy rate of the neural network with two hidden layers is significantly higher than the neural network with a single hidden layer, and the optimal number of neurons is 15.
  • the neural network decision formula in S40 is as follows:
  • X input is the input eigenvalue matrix
  • W 1 , W 2 , and W 3 are the weight matrices of each layer of the trained neural network
  • B 1 , B 2 , and B 3 are the bias matrices of each layer of the trained neural network.
  • activeFcn is the activation function and Y out is the network calculation result.
  • x is the input to the activation function and i is the ambient sound category.
  • the comprehensive decision-making module After receiving the classification results of the neural network classification module, the comprehensive decision-making module comprehensively analyzes a series of factors, mainly including the recognition results of the neural network and the size of the sound energy for a short period of time, gives a judgment of the current scene, and outputs the judgment results to Speech processing selection module.
  • the speech processing selection module selects the optimal speech processing program and its parameter configuration according to the judgment result of the current scene by the comprehensive decision module.
  • the eigenvalue selection uses a comprehensive analysis of the eigenvalue statistics and Gaussian mixture model, the average influence value algorithm, the sequence forward selection algorithm, and the method of classifier training result evaluation.
  • the amount of calculation of the eigenvalues and the calculation of the neural network does not exceed 20% of the calculation capacity of the cochlear speech processor.

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Abstract

An artificial cochlea ambient sound sensing method and system. The method comprises the following steps: a sound acquisition module acquires ambient sound in real time by using a microphone, and outputs an acquired discrete sound signal to a sound feature extraction module (S10); the sound feature extraction module processes sound signals transmitted by the sound acquisition module, extracts one group of feature values representing sound signal characteristics, and outputs the feature values to a neural network classification module (S20); the neural network classification module performs classification on the group of feature values by means of a trained neural network after one group of feature values extracted by the sound feature extraction module are received, and then outputs the classification result to an integrated decision module (S30); the integrated decision module integrally analyzes and determines the current scene after the classification result of the neural network classification module is received, and outputs the determination result to a voice processing selection module (S40); the voice processing selection module selects the optimal voice processing program and parameter configuration thereof according to the determination result of the integrated decision module for the current scene (S50).

Description

一种人工耳蜗的环境声感知方法和系统Ambient sound sensing method and system of cochlear implant 技术领域Technical field
本发明属于信号处理领域,涉及一种人工耳蜗的环境声感知方法和系统。The invention belongs to the field of signal processing, and relates to a method and system for sensing ambient sound of an artificial cochlea.
背景技术Background technique
人工耳蜗是目前市场上唯一一种可以有效地让重度或极重度耳聋患者恢复听力的医疗器械。一般的人工耳蜗的工作原理是将麦克风采集的声音信号,经过信号处理单元,将声音信号转为刺激编码发送给植入体,植入体按照刺激编码,通过微电极刺激听神经,从而使植入者恢复听力。跟助听器等其他辅听设备一样,这类系统缺少常人听力系统的一项重要功能,即可以在复杂的声音场景中分辨目标信号,并将其提取出来。比如在一群人或者相对嘈杂的环境下,听清聊天对象所说的话。通常的解决办法是通过一定的去噪算法,减少噪音对听声的影响。然而不同的环境(比如纯语音、待噪语音或者噪音环境)下的去噪算法和算法的参数配置是不同的。Cochlear implants are currently the only medical device on the market that can effectively restore hearing in patients with severe or severe deafness. The general working principle of a cochlear implant is to convert the sound signal collected by the microphone into a stimulus code and send it to the implant through the signal processing unit. The implant then stimulates the auditory nerve through microelectrodes according to the stimulus code, so that the implant Restore hearing. Like other hearing devices such as hearing aids, this type of system lacks an important function of ordinary human hearing systems, which is to distinguish target signals in complex sound scenes and extract them. For example, in a group of people or in a relatively noisy environment, listen to what the chat object is saying. The usual solution is to reduce the effect of noise on the sound of hearing through a certain denoising algorithm. However, the denoising algorithm and the parameter configuration of the algorithm are different in different environments (such as pure speech, speech to be noisy, or noise environment).
为了解决这类问题,又引入了环境声感知算法,系统可以根据环境声感知算法的判定结果,有针对性的开启降噪算法并配置相关参数。早期的人工耳蜗或者助听器的系统中,环境声感知算法的分类器采用的是隐马尔可夫模型。该模型相对简单,理论成熟较早,对训练数据要求不高,也保有一定的正确识别率。并且其运算量比较低,能够适应人工耳蜗这种运算能力有限的设备。随着近几年模式识别,机器学习等领域的不断创新,算力算法上的不断进步,更多的分类算法(支持向量机、神经网络等)在环境声感知领域上的表现更为突出,分类正确率更高。而且支持向量机、神经网络这类分类器相对于隐马尔可夫模型,将重心放在区别类别上,而无须提供类别转换的先验概率。也就是说只需要分析不同环境声的数据,不需要考虑从一种环境声转换另一种环境声的概率 是多少。得到这种转换概率非常困难,而且从数据上分析又不够准确。但是神经网络的变化很多,根据输入特征值的数目,隐含层的数目,每层网络节点数目的不同,其网络结构可以有很多种组合。而且通常神经网络的分类正确率跟其规模成正比,因而所需运算量也比较大。In order to solve this kind of problem, the ambient sound perception algorithm is also introduced. According to the judgment result of the ambient sound perception algorithm, the system can open the noise reduction algorithm and configure related parameters in a targeted manner. In early cochlear implants or hearing aid systems, the classifier of the ambient sound perception algorithm used a hidden Markov model. The model is relatively simple, the theory matures earlier, and it does not require high training data. It also maintains a certain correct recognition rate. And its calculation volume is relatively low, and it can adapt to such a device with limited computing power as a cochlear implant. With the continuous innovation of pattern recognition, machine learning and other fields in recent years, and the continuous improvement of computing power algorithms, more classification algorithms (support vector machines, neural networks, etc.) have become more prominent in the field of ambient sound perception. Higher classification accuracy. In addition, classifiers such as support vector machines and neural networks focus on distinguishing categories compared to hidden Markov models without providing prior probabilities of class conversion. In other words, only the data of different ambient sounds need to be analyzed, and the probability of converting one ambient sound to another ambient sound need not be considered. Obtaining this conversion probability is very difficult, and the analysis from the data is not accurate enough. But neural networks have many changes. According to the number of input feature values, the number of hidden layers, and the number of network nodes in each layer, the network structure can have many combinations. Moreover, the classification accuracy rate of neural networks is usually proportional to its scale, so the amount of computation required is also relatively large.
发明内容Summary of the invention
为解决上述问题,本发明针对现有声音感知处理的缺点,提出了一种人工耳蜗的环境声感知方法,采用神经网络,对环境声分类,该神经网络的输入特征值、网络结构是在人工耳蜗系统上进行的优化,即在满足一定的分类正确率的情况下,使运算量达到最小。In order to solve the above-mentioned problems, the present invention proposes an environmental sound sensing method of an artificial cochlea in response to the shortcomings of the existing sound perception processing. The neural sound network is used to classify the environmental sound. The optimization on the cochlear system is to minimize the amount of calculations when a certain classification accuracy is met.
为实现上述目的,本发明的技术方案为一种人工耳蜗的环境声感知方法,包括以下步骤:In order to achieve the above object, the technical solution of the present invention is a method for sensing an ambient sound of a cochlear implant, which includes the following steps:
声音采集模块采用麦克风实时采集环境声,然后将采集到的一段离散声音信号输出给声音特征提取模块;The sound collection module uses a microphone to collect the ambient sound in real time, and then outputs the collected discrete sound signal to the sound feature extraction module;
声音特征提取模块将声音采集模块发送来的声音信号作处理,提取一组代表声音信号特点的特征值,输出给神经网络分类模块;The sound feature extraction module processes the sound signals sent by the sound acquisition module, extracts a set of feature values representing the characteristics of the sound signals, and outputs them to the neural network classification module;
神经网络分类模块在收到声音特征提取模块提取的一组特征值之后,通过训练好的神经网络对该组特征值进行分类,然后将分类结果输出给综合决策模块;After receiving a set of feature values extracted by the sound feature extraction module, the neural network classification module classifies the set of feature values through a trained neural network, and then outputs the classification result to the comprehensive decision module;
综合决策模块在收到神经网络分类模块的分类结果之后,综合分析给出当前场景的判定,并将判定结果输出给语音处理选择模块;After receiving the classification result of the neural network classification module, the comprehensive decision module comprehensively analyzes and gives the judgment of the current scene, and outputs the judgment result to the speech processing selection module;
语音处理选择模块根据综合决策模块对当前场景的判定结果,选择最优的语音处理程序及其参数配置。The speech processing selection module selects the optimal speech processing program and its parameter configuration according to the judgment result of the current scene by the comprehensive decision module.
优选地,所述麦克风实时采集环境声使用全向麦克风或者麦克风阵列。Preferably, the microphone collects ambient sounds in real time using an omnidirectional microphone or a microphone array.
优选地,所述声音采集模块的采样率为16K。Preferably, the sampling rate of the sound acquisition module is 16K.
优选地,所述提取一组代表声音信号特点的特征值提取的特征值在8个。Preferably, the feature value extracted by extracting a set of feature values representing the characteristics of the sound signal is eight.
优选地,所述神经网络分类模块采用包含两个隐含层、每层15个神经元的 深度神经网络或者延迟神经网络。Preferably, the neural network classification module uses a deep neural network or a delayed neural network including two hidden layers and 15 neurons in each layer.
优选地,8个所述特征值从60个特征值中筛选而来。Preferably, the eight characteristic values are selected from 60 characteristic values.
优选地,所述特征值筛选采用综合分析特征值的统计值和高斯混合模型、平均影响值算法、序列前向选择算法、以及分类器训练结果评估的方法。Preferably, the feature value screening adopts a method of comprehensively analyzing the statistical value of the feature value and the Gaussian mixture model, the average influence value algorithm, the sequence forward selection algorithm, and the evaluation of the classifier training result.
优选地,所述特征值的计算量和神经网络的计算量不超过人工耳蜗言语处理器运算能力的20%。Preferably, the calculation amount of the characteristic value and the calculation amount of the neural network does not exceed 20% of the calculation capacity of the cochlear speech processor.
基于上述目的,本发明还提供了一种人工耳蜗的环境声感知系统,包括依次连接的声音采集模块、声音特征提取模块、神经网络分类模块、综合决策模块、语音处理选择模块,其中,Based on the above objectives, the present invention also provides an environmental sound perception system for a cochlear implant, which includes a sound acquisition module, a sound feature extraction module, a neural network classification module, a comprehensive decision module, and a speech processing selection module, which are connected in this order.
所述声音采集模块,用于采用麦克风实时采集环境声,然后将采集到的一段离散声音信号输出给声音特征提取模块;The sound collection module is configured to use a microphone to collect ambient sounds in real time, and then output the collected discrete sound signal to a sound feature extraction module;
所述声音特征提取模块,用于将声音采集模块发送来的声音信号作处理,提取一组代表声音信号特点的特征值,输出给神经网络分类模块;The sound feature extraction module is configured to process the sound signals sent by the sound acquisition module, extract a set of feature values representing the characteristics of the sound signal, and output to a neural network classification module;
所述神经网络分类模块,用于在收到声音特征提取模块提取的一组特征值之后,通过训练好的神经网络对该组特征值进行分类,然后将分类结果输出给综合决策模块;The neural network classification module is configured to, after receiving a set of feature values extracted by the sound feature extraction module, classify the set of feature values through a trained neural network, and then output the classification result to a comprehensive decision module;
所述综合决策模块,用于在收到神经网络分类模块的分类结果之后,综合分析给出当前场景的判定,并将判定结果输出给语音处理选择模块;The comprehensive decision module is configured to comprehensively analyze and give a judgment of the current scene after receiving the classification result of the neural network classification module, and output the judgment result to the voice processing selection module;
所述语音处理选择模块,用于根据综合决策模块对当前场景的判定结果,选择最优的语音处理程序及其参数配置。The speech processing selection module is configured to select an optimal speech processing program and its parameter configuration according to the determination result of the current scenario by the comprehensive decision module.
附图说明BRIEF DESCRIPTION OF THE DRAWINGS
图1为本发明实施例的人工耳蜗的环境声感知方法的步骤流程图;FIG. 1 is a flowchart of steps in a method for sensing an environmental sound of a cochlear implant according to an embodiment of the present invention; FIG.
图2为本发明实施例的人工耳蜗的环境声感知系统的结构框图;FIG. 2 is a structural block diagram of an environmental sound sensing system of a cochlear implant according to an embodiment of the present invention; FIG.
图3为本发明实施例的人工耳蜗的环境声感知方法和系统的神经网络分类模块具体示意图;FIG. 3 is a specific schematic diagram of a neural network classification module of a method and a system for sensing an environmental sound of a cochlear implant according to an embodiment of the present invention; FIG.
图4为本发明实施例的人工耳蜗的环境声感知方法的对不同隐含层和不同神经元数目的网络的运算量和正确率的对比图。FIG. 4 is a comparison diagram of a calculation amount and an accuracy rate of a method of sensing an ambient sound of a cochlear implant on networks of different hidden layers and different numbers of neurons according to an embodiment of the present invention.
具体实施方式detailed description
为了使本发明的目的、技术方案及优点更加清楚明白,以下结合附图及实施例,对本发明进行进一步详细说明。应当理解,此处所描述的具体实施例仅仅用以解释本发明,并不用于限定本发明。In order to make the objectives, technical solutions, and advantages of the present invention clearer, the present invention is further described in detail below with reference to the accompanying drawings and embodiments. It should be understood that the specific embodiments described herein are only used to explain the present invention and are not intended to limit the present invention.
相反,本发明涵盖任何由权利要求定义的在本发明的精髓和范围上做的替代、修改、等效方法以及方案。进一步,为了使公众对本发明有更好的了解,在下文对本发明的细节描述中,详尽描述了一些特定的细节部分。对本领域技术人员来说没有这些细节部分的描述也可以完全理解本发明。On the contrary, the invention covers any alternatives, modifications, equivalent methods, and schemes made on the spirit and scope of the invention as defined by the claims. Further, in order to give the public a better understanding of the present invention, in the following detailed description of the present invention, some specific details are described in detail. To those skilled in the art, the present invention can be fully understood without the description of these details.
参见图1,为本发明实施例的本发明的技术方案为人工耳蜗的环境声感知方法的步骤流程图,包括以下步骤:Referring to FIG. 1, a flowchart of steps of a method for sensing an environmental sound of a cochlear implant according to an embodiment of the present invention is provided, including the following steps:
S10,声音采集模块采用麦克风实时采集环境声,然后将采集到的一段离散声音信号输出给声音特征提取模块;S10: The sound collection module uses a microphone to collect ambient sounds in real time, and then outputs the collected discrete sound signal to the sound feature extraction module;
S20,声音特征提取模块将声音采集模块发送来的声音信号作处理,提取一组代表声音信号特点的特征值,输出给神经网络分类模块;S20: The sound feature extraction module processes the sound signals sent by the sound acquisition module, extracts a set of feature values representing the characteristics of the sound signal, and outputs them to the neural network classification module;
S30,神经网络分类模块在收到声音特征提取模块提取的一组特征值之后,通过训练好的神经网络对该组特征值进行分类,然后将分类结果输出给综合决策模块;S30. After receiving a set of feature values extracted by the sound feature extraction module, the neural network classification module classifies the set of feature values through a trained neural network, and then outputs the classification result to the comprehensive decision module;
S40,综合决策模块在收到神经网络分类模块的分类结果之后,综合分析给出当前场景的判定,并将判定结果输出给语音处理选择模块;S40. After receiving the classification result of the neural network classification module, the comprehensive decision module comprehensively analyzes and gives a judgment of the current scene, and outputs the judgment result to the voice processing selection module;
S50,语音处理选择模块根据综合决策模块对当前场景的判定结果,选择最优的语音处理程序及其参数配置。S50. The voice processing selection module selects an optimal voice processing program and its parameter configuration according to the judgment result of the current scenario by the comprehensive decision module.
本发明的系统实施例参见图2,包括依次连接的声音采集模块10、声音特征提取模块20、神经网络分类模块30、综合决策模块40、语音处理选择模块 50,其中,An embodiment of the system of the present invention is shown in FIG. 2 and includes a sound collection module 10, a sound feature extraction module 20, a neural network classification module 30, a comprehensive decision module 40, and a voice processing selection module 50, which are connected in this order.
声音采集模块10,用于采用麦克风实时采集环境声,然后将采集到的一段离散声音信号输出给声音特征提取模块20;A sound collection module 10, configured to use a microphone to collect ambient sounds in real time, and then output the collected discrete sound signal to the sound feature extraction module 20;
声音特征提取模块20,用于将声音采集模块发送来的声音信号作处理,提取一组代表声音信号特点的特征值,输出给神经网络分类模块30;The sound feature extraction module 20 is configured to process the sound signal sent by the sound acquisition module, extract a set of feature values representing the characteristics of the sound signal, and output it to the neural network classification module 30;
神经网络分类模块30,用于在收到声音特征提取模块提取的一组特征值之后,通过训练好的神经网络对该组特征值进行分类,然后将分类结果输出给综合决策模块40;A neural network classification module 30 is configured to classify the set of feature values through a trained neural network after receiving a set of feature values extracted by the sound feature extraction module, and then output the classification result to the comprehensive decision module 40;
综合决策模块40,用于在收到神经网络分类模块的分类结果之后,综合分析给出当前场景的判定,并将判定结果输出给语音处理选择模块50;The comprehensive decision module 40 is configured to, after receiving the classification result of the neural network classification module, comprehensively give a determination of the current scene, and output the determination result to the speech processing selection module 50;
语音处理选择模块50,用于根据综合决策模块对当前场景的判定结果,选择最优的语音处理程序及其参数配置。The voice processing selection module 50 is configured to select an optimal voice processing program and its parameter configuration according to the judgment result of the current scene by the comprehensive decision module.
具体实施例中,S10中麦克风实时采集环境声使用全向麦克风或者麦克风阵列,声音采集模块10的采样率为16K。In a specific embodiment, an omnidirectional microphone or a microphone array is used by the microphone to collect ambient sounds in real time in S10, and the sampling rate of the sound collection module 10 is 16K.
S20中提取一组代表声音信号特点的特征值提取的特征值在8个,8个特征值从60个特征值中筛选而来。在提取特征值之前做归一化处理,公式如下:A set of feature values representing the characteristics of the sound signal is extracted in S20. The extracted feature values are eight, and the eight feature values are filtered from the 60 feature values. The normalization process is performed before the feature values are extracted. The formula is as follows:
Figure PCTCN2019096648-appb-000001
Figure PCTCN2019096648-appb-000001
其中,x norm为归一化结果,X max为该特征值所在训练样本最大值,X min为该特征值所在训练样本最小值。 Among them, x norm is the normalized result, X max is the maximum value of the training sample where the feature value is, and X min is the minimum value of the training sample where the feature value is.
S30中神经网络分类模块采用包含两个隐含层、每层15个神经元的深度神经网络或者延迟神经网络。神经网络模块是通过大量数据样本训练得到的,以判别4类环境声(纯语音、带噪语音、噪音、音乐和安静)为例,其神经网络模 型参见图3。特征值选取1、2、3、4、5和6,一共六种组成一组。训练样本是从大量收集的音频文件中提取出来,一共包含了144000组样本特征值,每类环境声包含36000组特征值。为了找到运算量与正确率的平衡点,参见图4,我们尝试了1隐含层和2隐含层,每层不同神经元数目。从图中可以看出,两隐含层的神经网络的正确率明显高于单隐含层的神经网络,最佳神经元数目为15。The neural network classification module in S30 uses a deep neural network or a delayed neural network with two hidden layers and 15 neurons in each layer. The neural network module is obtained through training on a large number of data samples. Taking the discrimination of 4 types of environmental sounds (pure speech, noisy speech, noise, music, and quietness) as examples, see Figure 3 for its neural network model. The eigenvalues are selected from 1, 2, 3, 4, 5, and 6, and a total of six types form a group. The training samples are extracted from a large number of collected audio files, and they contain a total of 144,000 sets of sample feature values, and each type of ambient sound contains 36,000 sets of feature values. In order to find the balance between the amount of computation and the accuracy, see Figure 4, we tried 1 hidden layer and 2 hidden layers, each layer has a different number of neurons. It can be seen from the figure that the accuracy rate of the neural network with two hidden layers is significantly higher than the neural network with a single hidden layer, and the optimal number of neurons is 15.
S40中神经网络判定公式如下:The neural network decision formula in S40 is as follows:
Figure PCTCN2019096648-appb-000002
Figure PCTCN2019096648-appb-000002
其中,X input为输入特征值矩阵,W 1、W 2、W 3为训练好的神经网络每层权值矩阵,B 1、B 2、B 3为训练好的神经网络每层偏置矩阵,activeFcn为激活函数,Y out为网络计算结果。 Among them, X input is the input eigenvalue matrix, W 1 , W 2 , and W 3 are the weight matrices of each layer of the trained neural network, and B 1 , B 2 , and B 3 are the bias matrices of each layer of the trained neural network. activeFcn is the activation function and Y out is the network calculation result.
为了减少运算量,我们将隐含层的激活函数activeFcn H和输出层的激活函数activeFcn O分别定义为: To reduce the amount of computation, we define the activation function activeFcn H of the hidden layer and the activation function activeFcn O of the output layer as:
Figure PCTCN2019096648-appb-000003
Figure PCTCN2019096648-appb-000003
其中,x为激活函数的输入,i为环境声类别。Where x is the input to the activation function and i is the ambient sound category.
综合决策模块在收到神经网络分类模块的分类结果之后,综合分析一系列因素,主要包括一小段时间内神经网络的识别结果和声音能量大小,给出当前场景的判定,并将判定结果输出给语音处理选择模块。After receiving the classification results of the neural network classification module, the comprehensive decision-making module comprehensively analyzes a series of factors, mainly including the recognition results of the neural network and the size of the sound energy for a short period of time, gives a judgment of the current scene, and outputs the judgment results to Speech processing selection module.
语音处理选择模块根据综合决策模块对当前场景的判定结果,选择最优的语音处理程序及其参数配置。The speech processing selection module selects the optimal speech processing program and its parameter configuration according to the judgment result of the current scene by the comprehensive decision module.
特征值筛选采用综合分析特征值的统计值和高斯混合模型、平均影响值算法、序列前向选择算法、以及分类器训练结果评估的方法。The eigenvalue selection uses a comprehensive analysis of the eigenvalue statistics and Gaussian mixture model, the average influence value algorithm, the sequence forward selection algorithm, and the method of classifier training result evaluation.
特征值的计算量和神经网络的计算量不超过人工耳蜗言语处理器运算能力的20%。The amount of calculation of the eigenvalues and the calculation of the neural network does not exceed 20% of the calculation capacity of the cochlear speech processor.
以上所述仅为本发明的较佳实施例而已,并不用以限制本发明,凡在本发明的精神和原则之内所作的任何修改、等同替换和改进等,均应包含在本发明的保护范围之内。The above description is only the preferred embodiments of the present invention, and is not intended to limit the present invention. Any modification, equivalent replacement, and improvement made within the spirit and principle of the present invention shall be included in the protection of the present invention. Within range.

Claims (9)

  1. 一种人工耳蜗的环境声感知方法,其特征在于,包括以下步骤:An environmental sound sensing method of a cochlear implant, which comprises the following steps:
    声音采集模块采用麦克风实时采集环境声,然后将采集到的一段离散声音信号输出给声音特征提取模块;The sound collection module uses a microphone to collect the ambient sound in real time, and then outputs the collected discrete sound signal to the sound feature extraction module;
    声音特征提取模块将声音采集模块发送来的声音信号作处理,提取一组代表声音信号特点的特征值,输出给神经网络分类模块;The sound feature extraction module processes the sound signals sent by the sound acquisition module, extracts a set of feature values representing the characteristics of the sound signals, and outputs them to the neural network classification module;
    神经网络分类模块在收到声音特征提取模块提取的一组特征值之后,通过训练好的神经网络对该组特征值进行分类,然后将分类结果输出给综合决策模块;After receiving a set of feature values extracted by the sound feature extraction module, the neural network classification module classifies the set of feature values through a trained neural network, and then outputs the classification result to the comprehensive decision module;
    综合决策模块在收到神经网络分类模块的分类结果之后,综合分析给出当前场景的判定,并将判定结果输出给语音处理选择模块;After receiving the classification result of the neural network classification module, the comprehensive decision module comprehensively analyzes and gives the judgment of the current scene, and outputs the judgment result to the speech processing selection module;
    语音处理选择模块根据综合决策模块对当前场景的判定结果,选择最优的语音处理程序及其参数配置。The speech processing selection module selects the optimal speech processing program and its parameter configuration according to the judgment result of the current scene by the comprehensive decision module.
  2. 根据权利要求1所述的方法,其特征在于,所述麦克风实时采集环境声使用全向麦克风或者麦克风阵列。The method according to claim 1, wherein the microphone collects ambient sounds in real time using an omnidirectional microphone or a microphone array.
  3. 根据权利要求1所述的方法,其特征在于,所述声音采集模块的采样率为16K。The method according to claim 1, wherein a sampling rate of the sound acquisition module is 16K.
  4. 根据权利要求1所述的方法,其特征在于,所述提取一组代表声音信号特点的特征值提取的特征值在8个。The method according to claim 1, wherein the feature value extracted by extracting a set of feature values representing the characteristics of the sound signal is eight.
  5. 根据权利要求1所述的方法,其特征在于,所述神经网络分类模块采用包含两个隐含层、每层15个神经元的深度神经网络或者延迟神经网络。The method according to claim 1, wherein the neural network classification module uses a deep neural network or a delayed neural network including two hidden layers and 15 neurons in each layer.
  6. 根据权利要求4所述的方法,其特征在于,8个所述特征值从60个特征值中筛选而来。The method according to claim 4, wherein eight of the feature values are filtered from 60 feature values.
  7. 根据权利要求6所述的方法,其特征在于,所述特征值筛选采用综合分析特征值的统计值和高斯混合模型、平均影响值算法、序列前向选择算法、以及分类器训练结果评估的方法。The method according to claim 6, characterized in that the feature value screening adopts a method of comprehensively analyzing statistical values and Gaussian mixture models of feature values, an average influence value algorithm, a sequence forward selection algorithm, and a method for evaluating classifier training results .
  8. 根据权利要求1所述的方法,其特征在于,所述特征值的计算量和神经网络的计算量不超过人工耳蜗言语处理器运算能力的20%。The method according to claim 1, wherein the calculation amount of the characteristic value and the calculation amount of the neural network does not exceed 20% of the calculation capacity of the cochlear speech processor.
  9. 采用权利要求1-8之一所述方法的系统,其特征在于,包括依次连接的声音采集模块、声音特征提取模块、神经网络分类模块、综合决策模块、语音处理选择模块,其中,The system adopting the method according to any one of claims 1 to 8, further comprising a sound acquisition module, a sound feature extraction module, a neural network classification module, a comprehensive decision module, and a speech processing selection module connected in sequence, wherein:
    所述声音采集模块,用于采用麦克风实时采集环境声,然后将采集到的一段离散声音信号输出给声音特征提取模块;The sound collection module is configured to use a microphone to collect ambient sounds in real time, and then output the collected discrete sound signal to a sound feature extraction module;
    所述声音特征提取模块,用于将声音采集模块发送来的声音信号作处理,提取一组代表声音信号特点的特征值,输出给神经网络分类模块;The sound feature extraction module is configured to process the sound signals sent by the sound acquisition module, extract a set of feature values representing the characteristics of the sound signal, and output to a neural network classification module;
    所述神经网络分类模块,用于在收到声音特征提取模块提取的一组特征值之后,通过训练好的神经网络对该组特征值进行分类,然后将分类结果输出给综合决策模块;The neural network classification module is configured to, after receiving a set of feature values extracted by the sound feature extraction module, classify the set of feature values through a trained neural network, and then output the classification result to a comprehensive decision module;
    所述综合决策模块,用于在收到神经网络分类模块的分类结果之后,综合分析给出当前场景的判定,并将判定结果输出给语音处理选择模块;The comprehensive decision module is configured to comprehensively analyze and give a judgment of the current scene after receiving the classification result of the neural network classification module, and output the judgment result to the voice processing selection module;
    所述语音处理选择模块,用于根据综合决策模块对当前场景的判定结果,选择最优的语音处理程序及其参数配置。The speech processing selection module is configured to select an optimal speech processing program and its parameter configuration according to the determination result of the current scenario by the comprehensive decision module.
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