# English Speech Recognition System Model Based on Computer-Aided Function and Neural Network Algorithm

Authors: Jin Zhang
Journal: Computational Intelligence and Neuroscience
 (2022)

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## Abstract

With the economic globalization continuous growth of China’s socioeconomic level tends to be internationalized, China’s attention to English has been significantly improved. However, the domestic English teaching level is limited, so it is impossible to correct students’ English pronunciation and make a reasonable evaluation at all times so that oral training has certain disadvantages. However, the computer-aided language learning system at home and abroad focuses on the practice of words and grammar, and the evaluation indicators are less and not comprehensive. In view of the complexity of English pronunciation changes, traditional speech recognition is difficult to recognize speech speed and improve its accuracy. Furthermore, to strengthen the English pronunciation of domestic students, a nonlinear network structure is studied in depth to simulate the human brain to analyze a model of speech recognition is established Mel frequency cepstrum characteristic parameters of human ear model and deep belief network. In this paper, the traditional computer pronunciation evaluation method is improved in an all-round way, and a set of high-quality speech recognition system of speech recognition method is constructed. Aiming at the above problems, it takes the students as the research, which proves that the method adopted in this paper can give the learners accurate pronunciation quality analysis report and guidance and correct their intonation and improve the learning effect, and the experimental data verify that the improved speech recognition system model recognition ability is higher than the traditional model.


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## Body

## 1. Introduction

With the increase of globalization and China’s internationalization, the Chinese people’s demand for learning English is also easy to grow. However, because the Chinese vocal characteristics are different from those of the British, China does not have an English learning environment and lacks excellent English teachers. Due to the influence of time, location, and other factors, traditional classroom teaching can no longer meet the needs of English learning. At present, there are still great problems in English teaching; for example, teachers’ descriptions are too general, and students’ subjective initiative in learning cannot be fully brought into play, and learning English has become one of the research hotspots in the field of education. With the development of computer, the progress of language teaching methods, computer-aided language learning [1] provides the possibility to solve this problem. This technology allows learners to study in no time or area, can give learners timely and sharp evaluation and guidance, and can also help learners to find differences and standard pronunciation in different ways. Allowing learners to listen repeatedly, compare and correct pronunciation errors, and improve language learning efficiency.The research goal of the call is to combine computer technology, especially speech recognition and evaluation technology, with the current language teaching methods to further establish a call system. Some famous international university research institutions are carrying out relevant research, such as Carnegie Mellon University in the United States, Tokyo University in Japan, and Cambridge University in the United Kingdom. MIT and Cambridge are working together to develop the oral communicative language learning project (SCILL), which aims to develop a conversational system to provide learners with conversational practice. WebGrader developed by Stanford University uses a speech recognition engine [2] and scoring technology to score pronunciation exercises for second language learners. This conversational system allows students to practice spoken English online and promotes communication in English learning. In the research of speech quality evaluation in China, the Institute of automation, Tsinghua University, Microsoft Asia Research Institute, and other institutions are carrying out relevant content research and have achieved good results. The core of call is high-precision speech recognition is the important foundation of speech evaluation. DTW, HMM, and ANN are the classical mainstream methods, each with its advantages and disadvantages.In recent years, the development of DL [3], BD [4], and CCT [5] and quality assessment technology have made rapid progress. DL originated from the field of ML [6], which aims to establish and simulate the deep neural network (DNN) of the human brain for analysis and learning. DNN can multilayer deep transmission of data by human brain neurons. Therefore, the study of English SRT based on DL can greatly improve the ability of voice aspect, improve access to information, and improve the user experience.At present, with the rise of the international market, the training of spoken English in colleges and universities has been continuously strengthened. Therefore, the following problems are encountered in the study of spoken: Most people still choose language transponder and MP3 player, to learn oral English. However, none of these methods can carry out speech recognition, and they are limited to voice query and tracking functions. In addition, limited by technological factors, some call systems at home and abroad mainly concentrate on the learning of words and so on. Evaluation indicators are used as the evaluation, which can give an accurate score. Limited by their own level is not enough, and it is hard for them to find their mistakes and correct them with a total score of one point. In terms of oral test evaluation, the current oral test is still based on strong subjective will, different standards, and slow manual scoring. Due to the different experiences of scoring experts, and the different status of the same expert, the evaluation of the same voice will be biased. The English speech recognition system model constructed in this paper mainly uses the deep neural network to make the speech recognition can automatically recognize speech.In order to solve those shortcomings, the DL application is selected, and a speech recognition model based on DBN is constructed [7]. Speech quality has been improved, and a reasonable speech quality evaluation model has been established considering the multiparameter indexes and their weights such as voice tone, voice speed, rhythm, and voice tone. The original achievement of the research can be applied to the study and training of human-computer interaction oral English. It can give learners more intuitive and useful evaluation, give advice, and give wrong pronunciation. It can quickly guide and motivate learners to learn English. The results are helpful to solve the problems in oral English teaching. In addition, English STR can improve the articulation of speech in the aspect of speech transmission, and the collection of causes is also more clear. The pronunciation of each syllable in English can be clearly collected, and the sound quality and timbre of speech can be guaranteed in the process of transmission.


## 2. Speech Signal Preprocessing and Feature Extraction

### 2.1. Basic Principles of SRT

SRT is one of the most popular text captions because its simple specification makes it easy to create and modify. The purpose of SR [8] is to decode the feature stream M of speech recognition and get the corresponding word sequence N, that is, to find M=argmaxMpN/M. It includes the following key modules: feature extraction, acoustic model, speech dictionary, language model. In general, it is difficult to search the optimal word sequence directly through features, so the transformation is based on Bayesian criteria.(1)PM/N=PM/NPMPN,where pM/N is the generating acoustic feature M for a given word sequence N, which is the probability that the acoustic model needs to estimate. In the recognition model, if the speech features need to be recognized and extracted, the recognition program needs to be input into the system to complete the feature extraction. PN is the observed word sequence, which is the problem to be solved by the language model, and the construction of the speech recognition system is transformed into modular acoustic modeling, language modeling, word sequence decoding, and so on. The traditional speech recognition framework is shown in Figure 1.(1)
SSP and AFE: SSP [9] needs to preprocess and extract feature parameters. Generally speaking, the process of feature extraction is independent of the training of the recognition model and the testing of the system. According to the previous research experience, the feature parameter extraction of speech can be completed through a series of mathematical processes. The common speech feature parameters are linear prediction cepstrum coefficient and MFCC. MFCC parameters simulate the auditory characteristics of human ears. In most cases, MFCC is the most commonly used.(2)
Acoustic model [10]: HMM is a very popular acoustic model in speech recognition. The state jump model is consistent with the short-term stability of speech signals, so HMM can model continuous speech signals. In the GMM-HMM model, the probability of observation sequence is output by GMM, then the likelihood degree of observation sequence is obtained by HMM, and then, the optimal sequence path is obtained by decoding algorithm. With the continuous in-depth study of researchers, we also use DNN and HMM as acoustic models, in which DNN replaces GMM to generate the probability of observation traits, and HMM is used to describe the dynamic characteristics of speech signals. The model has been proved to be feasible and widely used in the speech recognition system.(3)
Language model [11]: M-Gram is most commonly used in speech recognition research. Language model based on RNN language model RNNLM is gradually introduced into speech recognition. This model can model long history information and has a better performance compared with n-gram. However, in large vocabulary speech recognition tasks, replacing n-gram with RNNLM will increase, and because of the calculation time, the recognition efficiency of the whole system will decrease.(4)
Decoding search: After the score of the acoustic model is obtained through the acoustic model, the language model is combined with the relevant algorithm to get the final recognition sequence. For example, the dimension bit algorithm is used to decode the HMM system, and some other models also use clustering search to obtain the optimal sequence.Figure 1
Structure of traditional speech recognition.


### 2.2. Speech Signal Preprocessing

Before speech signal preprocessing, it must be preweighted, framing, windowing, and endpoint detection. This is done to eliminate the influence of harmonics on the quality of speech signal, which is caused by human speech organs. SP affects the effect of speech, but a smoother speech signal can provide better parameters. Improving voice quality can solve coverage problems by reducing interference factors and improving the quality of the hardware.


### 2.3. Recognition of Speech Characteristic Parameters

Speech recognition is the technology or function that enables a program or system to process human speech. It is mainly used to convert spoken language into computer text. The purpose of speech recognition is to make the machine finally understand the oral nature of human language and the form of a voice in the digital signal of the computer. It is obvious that the digital signal directly understood by the computer will make the whole model calculation huge. The voice signal is converted into the dimension that can reduce the characteristic parameters of speech and, at the same time, ensure the maximum retention of useful information. Speech features have better recognition characteristics, which is convenient for the modeling of speech recognition system and has the certain anti-interference ability to the environment. MFCC [12] is a cepstrum feature parameter extracted based on the critical frequency band effect of the human ear. In many researches, MFCC parameters and their differences are usually used to model speech recognition tasks.Figure2 describes the extraction structure of MFCC feature parameters. A complete speech is divided into equal length and adjacent frames by framing. In order to make the overlapped part of the adjacent frame signal smooth transition, Hamming window function is used to window the signal. Then, the fast Fourier transform is used to transform. Then, the Mel triangle filter is used to calculate the signal, and then, the logarithm is taken to get the Mel spectrum of the signal. Finally, MFCC parameters are obtained by DCT. MFCC parameters reflect the static characteristics of speech. Usually, after getting the parameters of MFCC, the dynamic change of speech is described by calculating the difference, and the recognition performance of the system is improved by combining the static and dynamic methods. For example, MFCC can obtain the time length of speech and then divide it into N speech segments for recognition according to the length of the speech.Figure 2
MFCC parameter extraction process.


## 2.1. Basic Principles of SRT

SRT is one of the most popular text captions because its simple specification makes it easy to create and modify. The purpose of SR [8] is to decode the feature stream M of speech recognition and get the corresponding word sequence N, that is, to find M=argmaxMpN/M. It includes the following key modules: feature extraction, acoustic model, speech dictionary, language model. In general, it is difficult to search the optimal word sequence directly through features, so the transformation is based on Bayesian criteria.(1)PM/N=PM/NPMPN,where pM/N is the generating acoustic feature M for a given word sequence N, which is the probability that the acoustic model needs to estimate. In the recognition model, if the speech features need to be recognized and extracted, the recognition program needs to be input into the system to complete the feature extraction. PN is the observed word sequence, which is the problem to be solved by the language model, and the construction of the speech recognition system is transformed into modular acoustic modeling, language modeling, word sequence decoding, and so on. The traditional speech recognition framework is shown in Figure 1.(1)
SSP and AFE: SSP [9] needs to preprocess and extract feature parameters. Generally speaking, the process of feature extraction is independent of the training of the recognition model and the testing of the system. According to the previous research experience, the feature parameter extraction of speech can be completed through a series of mathematical processes. The common speech feature parameters are linear prediction cepstrum coefficient and MFCC. MFCC parameters simulate the auditory characteristics of human ears. In most cases, MFCC is the most commonly used.(2)
Acoustic model [10]: HMM is a very popular acoustic model in speech recognition. The state jump model is consistent with the short-term stability of speech signals, so HMM can model continuous speech signals. In the GMM-HMM model, the probability of observation sequence is output by GMM, then the likelihood degree of observation sequence is obtained by HMM, and then, the optimal sequence path is obtained by decoding algorithm. With the continuous in-depth study of researchers, we also use DNN and HMM as acoustic models, in which DNN replaces GMM to generate the probability of observation traits, and HMM is used to describe the dynamic characteristics of speech signals. The model has been proved to be feasible and widely used in the speech recognition system.(3)
Language model [11]: M-Gram is most commonly used in speech recognition research. Language model based on RNN language model RNNLM is gradually introduced into speech recognition. This model can model long history information and has a better performance compared with n-gram. However, in large vocabulary speech recognition tasks, replacing n-gram with RNNLM will increase, and because of the calculation time, the recognition efficiency of the whole system will decrease.(4)
Decoding search: After the score of the acoustic model is obtained through the acoustic model, the language model is combined with the relevant algorithm to get the final recognition sequence. For example, the dimension bit algorithm is used to decode the HMM system, and some other models also use clustering search to obtain the optimal sequence.Figure 1
Structure of traditional speech recognition.


## 2.2. Speech Signal Preprocessing

Before speech signal preprocessing, it must be preweighted, framing, windowing, and endpoint detection. This is done to eliminate the influence of harmonics on the quality of speech signal, which is caused by human speech organs. SP affects the effect of speech, but a smoother speech signal can provide better parameters. Improving voice quality can solve coverage problems by reducing interference factors and improving the quality of the hardware.


## 2.3. Recognition of Speech Characteristic Parameters

Speech recognition is the technology or function that enables a program or system to process human speech. It is mainly used to convert spoken language into computer text. The purpose of speech recognition is to make the machine finally understand the oral nature of human language and the form of a voice in the digital signal of the computer. It is obvious that the digital signal directly understood by the computer will make the whole model calculation huge. The voice signal is converted into the dimension that can reduce the characteristic parameters of speech and, at the same time, ensure the maximum retention of useful information. Speech features have better recognition characteristics, which is convenient for the modeling of speech recognition system and has the certain anti-interference ability to the environment. MFCC [12] is a cepstrum feature parameter extracted based on the critical frequency band effect of the human ear. In many researches, MFCC parameters and their differences are usually used to model speech recognition tasks.Figure2 describes the extraction structure of MFCC feature parameters. A complete speech is divided into equal length and adjacent frames by framing. In order to make the overlapped part of the adjacent frame signal smooth transition, Hamming window function is used to window the signal. Then, the fast Fourier transform is used to transform. Then, the Mel triangle filter is used to calculate the signal, and then, the logarithm is taken to get the Mel spectrum of the signal. Finally, MFCC parameters are obtained by DCT. MFCC parameters reflect the static characteristics of speech. Usually, after getting the parameters of MFCC, the dynamic change of speech is described by calculating the difference, and the recognition performance of the system is improved by combining the static and dynamic methods. For example, MFCC can obtain the time length of speech and then divide it into N speech segments for recognition according to the length of the speech.Figure 2
MFCC parameter extraction process.


## 3. Research on Speech Recognition Model

### 3.1. Concept of NN

NN [13] can be regarded as bionic because the neural network is designed according to the human brain nerve and have the function of calculation and decision-making. The neural network is mainly used in classifiers. The NN calculates layer by layer according to the input characteristic quantity and finally outputs the identified signal. The NN itself has the function of learning. In the training stage, it can learn the unique characteristics of different samples according to the training samples and carry out memory; in the test samples, when encountering similar samples, it can distinguish the type of samples according to “memory,” so the NN needs a large number of complete samples, and can correctly identify new samples through continuous learning and memory. It can ameliorate the generalization of NN. In the process of learning, if the training samples are not enough, it will lead to underfitting, and the actual classification effect will be very poor. If the training samples are single or overtrained, there will be overfitting. The overfitting network has a high recognition rate in training samples, excellent performance, and low recognition rate in test samples.Different types of neural networks have different application scenarios. Common neural networks include BP NN [14], convolution neural network [15], wavelet NN [16], radial basis network [17], and self-organizing NN [18], among which BP NN is mainly for classification; convolution NN is mainly used for image recognition. Although the neural network is widely used in engineering, such as fingerprint recognition, face recognition, and biological body recognition, it is still unable to discuss the specific learning logic in theory, such as how to learn and remember the weight or value according to a certain function change and how to improve the number of hidden layers of the NN. In engineering, the parameters of the neural network are mainly adjusted, and the best is determined by the method according to the results of identification. With the increase of engineering complexity, there will be more and more related parameters, and the task of related parameter adjustment will gradually increase.The artificial neuron [19] can be expressed by the following formula:(2)a=∑i=1rωixib=fa+θ.Here,x1,x2,…,xr represents the r input of the neuron; ωi represents the connection strength (connection weight) of the ith input; θ is the bias of the neuron; b is the neuron. Therefore, the artificial neuron is a multi-input single-output nonlinear structure.


### 3.2. BP Neural Network

Artificial NN, also known as NN, is a data processing model based on biological NN to each other for calculation and changes its structure according to external information, mainly through adjusting the weight of the network training input data and model, so that it has the ability to finally solve practical problems.At present, a variety of neural network models have been put forward in the academic field, including radial basis function network [20], Hopfield network [21], CMAC cerebellum model [22], and BP NN. This paper studies BP NN. BP NN is usually multilayer, and another related concept is MLP, which has multiple hidden layers. MLP emphasizes the structure of NN, while BP NN uses the BP algorithm to adjust the network weight on the premise of a multilayer network. In most cases, they usually refer to the same network.


### 3.3. Deep Learning NN

Deep NN is defined as a perceptron (MLP) with multiple layers that is usually greater than 2. Figure3 depicts a NN with one input layer, one output layer, and three hidden layers. Neurons between adjacent layers are connected by weight. Through the input v1, weight W1, and deviation b1 of this layer, the input of the next layer can be obtained by using the activation function.(3)vl+1=fvlWl+b1.Figure 3
Depth neural network model.Error backpropagation (BP) method is the deep NN. By calculating the partial derivative of each weight and the deviation of the loss function, the descent gradient of each parameter is obtained, and then, the parameter is updated by the gradient.The depth NN can be represented in many different ways by different structures of hidden layer elements and different connection ways between adjacent layers. In speech recognition tasks, RNN and its variant CBLSTM have achieved good results and have been widely used.The RNN connects the hidden layer elements of the feedforward neural network. When calculating the output of the hidden layer node, its calculation result depends on the calculation result of the previous node. Figure3 depicts the RNN structure:The hidden layer of RNN is self-connected to expand the hidden layer.U, V , and W are connection output layer and hidden layer, respectively. The connection weights of all nodes are shared. xt is the input of the current node, that is, the characteristic parameter vector obtained from the current voice frame signal, and the calculation formula of the hidden layer st and the output ot are as follows:(4)st=tanhxtU+xt−1W+bs,ot=stV.The advantage of RNN hidden layer self-connection is that by extending it, the network of any length in the sequence can be obtained. In addition, different speech lengths are inconsistent. It provides a very flexible input model for speech signal by using the hidden layer self-connection of RNN.


### 3.4. Research on Speech Recognition Based on DBN

The core of the deep belief network is a layer greedy learning algorithm and then uses the backpropagation algorithm to fine-tune the network to get a better performance network model. Experiments show that the weight initialization of the multilayer perceptron corresponds to this network model.Deep Boltzmann machines (DBMs) can be obtained by using an unsupervised greedy layer-by-layer approach. As shown in Figure4 (i.e., the Bayesian belief network does not connect the layers between nodes) to approach the visual layer, layer by layer error propagation, and contain the farthest part used from the visual layer, we can obtain the deep belief network speech recognition model (DBN). In essence, RBM training can obtain the global optimal initial parameters, so as to improve the network performance [23, 24].Figure 4
DBN voice model.


### 3.5. Content of Speech Recognition

As shown in Figure5 first of all, the voice analog signal is digitized, and the voice signal is collected by the sound card of PC. According to Nyquist sampling theorem, in the process of analog/digital signal conversion, when the sampling frequency fs_max is greater than twice the highest frequency of Fmax in the signal, the following formula is obtained:(5)fs_max≥2∗Fmax.Figure 5
Content of speech recognition.The sampled digital signal can fully express the information in the speech. The normal speech is generally 50∼5000 Hz, and this paper sets the sampling as B kHz including preweighting and window. Preprocessed speech signal was collected. The speech features that can be recognized by the recognition model include auditory features, because the ultimate purpose of speech is to be listened to by human beings, and speech also has specific timbre features. Finally, the selected speech feature is recognized by the speech recognition model designed in this paper.


## 3.1. Concept of NN

NN [13] can be regarded as bionic because the neural network is designed according to the human brain nerve and have the function of calculation and decision-making. The neural network is mainly used in classifiers. The NN calculates layer by layer according to the input characteristic quantity and finally outputs the identified signal. The NN itself has the function of learning. In the training stage, it can learn the unique characteristics of different samples according to the training samples and carry out memory; in the test samples, when encountering similar samples, it can distinguish the type of samples according to “memory,” so the NN needs a large number of complete samples, and can correctly identify new samples through continuous learning and memory. It can ameliorate the generalization of NN. In the process of learning, if the training samples are not enough, it will lead to underfitting, and the actual classification effect will be very poor. If the training samples are single or overtrained, there will be overfitting. The overfitting network has a high recognition rate in training samples, excellent performance, and low recognition rate in test samples.Different types of neural networks have different application scenarios. Common neural networks include BP NN [14], convolution neural network [15], wavelet NN [16], radial basis network [17], and self-organizing NN [18], among which BP NN is mainly for classification; convolution NN is mainly used for image recognition. Although the neural network is widely used in engineering, such as fingerprint recognition, face recognition, and biological body recognition, it is still unable to discuss the specific learning logic in theory, such as how to learn and remember the weight or value according to a certain function change and how to improve the number of hidden layers of the NN. In engineering, the parameters of the neural network are mainly adjusted, and the best is determined by the method according to the results of identification. With the increase of engineering complexity, there will be more and more related parameters, and the task of related parameter adjustment will gradually increase.The artificial neuron [19] can be expressed by the following formula:(2)a=∑i=1rωixib=fa+θ.Here,x1,x2,…,xr represents the r input of the neuron; ωi represents the connection strength (connection weight) of the ith input; θ is the bias of the neuron; b is the neuron. Therefore, the artificial neuron is a multi-input single-output nonlinear structure.


## 3.2. BP Neural Network

Artificial NN, also known as NN, is a data processing model based on biological NN to each other for calculation and changes its structure according to external information, mainly through adjusting the weight of the network training input data and model, so that it has the ability to finally solve practical problems.At present, a variety of neural network models have been put forward in the academic field, including radial basis function network [20], Hopfield network [21], CMAC cerebellum model [22], and BP NN. This paper studies BP NN. BP NN is usually multilayer, and another related concept is MLP, which has multiple hidden layers. MLP emphasizes the structure of NN, while BP NN uses the BP algorithm to adjust the network weight on the premise of a multilayer network. In most cases, they usually refer to the same network.


## 3.3. Deep Learning NN

Deep NN is defined as a perceptron (MLP) with multiple layers that is usually greater than 2. Figure3 depicts a NN with one input layer, one output layer, and three hidden layers. Neurons between adjacent layers are connected by weight. Through the input v1, weight W1, and deviation b1 of this layer, the input of the next layer can be obtained by using the activation function.(3)vl+1=fvlWl+b1.Figure 3
Depth neural network model.Error backpropagation (BP) method is the deep NN. By calculating the partial derivative of each weight and the deviation of the loss function, the descent gradient of each parameter is obtained, and then, the parameter is updated by the gradient.The depth NN can be represented in many different ways by different structures of hidden layer elements and different connection ways between adjacent layers. In speech recognition tasks, RNN and its variant CBLSTM have achieved good results and have been widely used.The RNN connects the hidden layer elements of the feedforward neural network. When calculating the output of the hidden layer node, its calculation result depends on the calculation result of the previous node. Figure3 depicts the RNN structure:The hidden layer of RNN is self-connected to expand the hidden layer.U, V , and W are connection output layer and hidden layer, respectively. The connection weights of all nodes are shared. xt is the input of the current node, that is, the characteristic parameter vector obtained from the current voice frame signal, and the calculation formula of the hidden layer st and the output ot are as follows:(4)st=tanhxtU+xt−1W+bs,ot=stV.The advantage of RNN hidden layer self-connection is that by extending it, the network of any length in the sequence can be obtained. In addition, different speech lengths are inconsistent. It provides a very flexible input model for speech signal by using the hidden layer self-connection of RNN.


## 3.4. Research on Speech Recognition Based on DBN

The core of the deep belief network is a layer greedy learning algorithm and then uses the backpropagation algorithm to fine-tune the network to get a better performance network model. Experiments show that the weight initialization of the multilayer perceptron corresponds to this network model.Deep Boltzmann machines (DBMs) can be obtained by using an unsupervised greedy layer-by-layer approach. As shown in Figure4 (i.e., the Bayesian belief network does not connect the layers between nodes) to approach the visual layer, layer by layer error propagation, and contain the farthest part used from the visual layer, we can obtain the deep belief network speech recognition model (DBN). In essence, RBM training can obtain the global optimal initial parameters, so as to improve the network performance [23, 24].Figure 4
DBN voice model.


## 3.5. Content of Speech Recognition

As shown in Figure5 first of all, the voice analog signal is digitized, and the voice signal is collected by the sound card of PC. According to Nyquist sampling theorem, in the process of analog/digital signal conversion, when the sampling frequency fs_max is greater than twice the highest frequency of Fmax in the signal, the following formula is obtained:(5)fs_max≥2∗Fmax.Figure 5
Content of speech recognition.The sampled digital signal can fully express the information in the speech. The normal speech is generally 50∼5000 Hz, and this paper sets the sampling as B kHz including preweighting and window. Preprocessed speech signal was collected. The speech features that can be recognized by the recognition model include auditory features, because the ultimate purpose of speech is to be listened to by human beings, and speech also has specific timbre features. Finally, the selected speech feature is recognized by the speech recognition model designed in this paper.


## 4. Evaluation of Multiparameter Pronunciation Quality

### 4.1. Process for Assessing Pronunciation Quality

The judge of voice quality is divided into subjective evaluation and objective evaluation, as shown in Figure6.Figure 6
A comparison between subjective and objective assessment of pronunciation quality.Objective evaluation is the prediction and estimation of subjective evaluation, so the accuracy of subjective evaluation directly affects the performance of objective evaluation. Only when the experts evaluate the speech quality of the test speech truly and reliably and, on this basis, carry out machine evaluation, the objective evaluation is more meaningful. In order to get a real and reliable expert score, it is necessary to establish an appropriate voice quality evaluation standard according to the application requirements.According to different situations, this paper uses two objective scoring methods to calculate scores.(1)
Quadratic curve fitting method is mainly used for the verification of single speech feature experiment. The specific process is as follows: the target distance between the training samples and the reference speech is calculated by using the training samples in the language library, and then, the parameters of the quadratic fitting curve are calculated by using the curve fitting method according to the principle of minimum variance. According to the parameters of the quadratic fitting curve, the subjective and objective mapping test is carried out, and the corresponding relationship between the objective distortion measurementDk and the subjective estimation MOS is obtained.(6)M=a·D2k+b·Dk+C1<k≤K,wherea, b, C are the parameters of the quadratic fitting curve obtained by training, M is the target distance value, Dk is the objective estimate of MOS subjective score obtained by the quadratic fitting curve, and K is the total number of distortion conditions.(2)
BP scoring method mainly aims at the overall evaluation system of this paper, that is, using the BP algorithm to calculate the objective score of test speech.


### 4.2. Quality Evaluation Index

In recent years, Chinese English learners have made a great breakthrough in the overall level of pronunciation, especially in the pronunciation of monosyllabics (vowel-consonant), such as vowels in the back and change. This is largely due to the country’s investment in educational resources and the enthusiasm of English learners. Therefore, for most English learners, they can better master the pronunciation of English words. But in real life, the rhythm of sentences is the key to English fluency. Many learners encounter a bottleneck after mastering the pronunciation of a single sound because they find that they can pronounce each sound very standard but still can express English with the strong Chinese flavor. This group includes not only beginners but also many intermediate-level learners, English majors, and even some people who use English for a long time. In a sentence, prosody is a very important factor. Every language has its own prosody.The evaluation of voice quality mainly includes voice tone, voice length, and voice rhythm. In the process of evaluation, phonemes and words are mainly evaluated by intonation. Sentences and paragraphs should not only check the content of the pronunciation itself but also determine the true meaning of the sentence to a large extent. When evaluating the pronunciation quality of sentences and paragraphs, we need to consider the prosody information comprehensively, such as whether the speaker can grasp the key information of the sentence more accurately, whether the sentence is not so important, and whether the length of the voice is appropriate. In other words, when evaluating the voice quality of English sentences, good voice quality not only requires complete, no pronunciation errors but also needs speaking speed, accent, and accuracy. Therefore, this paper uses intonation and prosody to evaluate the quality of speech.From the perspective of linguistics, prosody, also known as rhythm, refers to the organization of phonemes higher than the level of segments. It is a systematic organization that organizes various language units into discourse or related chunks in discourse. The realization of prosody can not only convey linguistic information but also paralinguistic and nonverbal information (including emphasis, emphasis, phrase grouping), as well as emotion, attitude, vocabulary discrimination, and other functions. Therefore, the quality of English pronunciation is closely related to its rhythm.Prosodic features, also known as suprasegmental features, mainly refer to speed, rhythm, and intonation composed of dynamic modes such as pitch, intensity, and duration. This paper uses speech speed evaluation and paired variation basic frequency intonation to evaluate and guide English speech quality.


### 4.3. Evaluation Model of Multiparameter Pronunciation Quality

Different groups (such as college students and business people) have different requirements for learning oral English. Based on the evaluation of voice tone, speed, rhythm, intonation, and other quality indicators achieved by taking oral English of college students as the research object, a comprehensive analysis of various indicators is made. In the comprehensive quality evaluation, the relationship between the weight of each index should be considered, and a multiparameter evaluation model and method of college students’ English pronunciation quality should be established to evaluate the pronunciation quality reasonably and objectively.


### 4.4. Speech Evaluation Experiment

The purpose of the speech evaluation experiment is to verify the performance of the English speech quality evaluation model and method proposed in this paper. In this paper, consistency rate and PCC are used to express the manual evaluation.The consistency ratio refers to the ratio of sample size, which is consistent with machine evaluation and nonautomatic evaluation. The specific calculation method is as follows:(7)ASame rate=Evaluate the same number of samplesTotal sample size.The adjacent consistency rate is the ratio between the manual evaluation and the sum of adjacent samples and total samples, in which “adjacent” is defined as one level difference between machine evaluation and manual evaluation. The specific calculation method is as follows:(8)AAdjacent consistency rate=Same number of samples+adjacent samplesTotal sample size.The two variables, which is expressed byr, are the linear correlation between two variables, and the value range is −1∼+1. The greater the absolute value, the stronger the correlation. Generally, 0<R<0.2 indicates very weak correlation, 0.2<R<0.4 indicates very weak correlation, 0.4<R<0.6 indicates medium correlation, 0.6<R<0.8 indicates very strong correlation, and 0.8<R<1 indicates very strong correlation. The specific calculation method is as follows:(9)r=N∑xiyi−∑xi∑yiN∑xi2−∑xi2N∑yi2−∑yi2.


## 4.1. Process for Assessing Pronunciation Quality

The judge of voice quality is divided into subjective evaluation and objective evaluation, as shown in Figure6.Figure 6
A comparison between subjective and objective assessment of pronunciation quality.Objective evaluation is the prediction and estimation of subjective evaluation, so the accuracy of subjective evaluation directly affects the performance of objective evaluation. Only when the experts evaluate the speech quality of the test speech truly and reliably and, on this basis, carry out machine evaluation, the objective evaluation is more meaningful. In order to get a real and reliable expert score, it is necessary to establish an appropriate voice quality evaluation standard according to the application requirements.According to different situations, this paper uses two objective scoring methods to calculate scores.(1)
Quadratic curve fitting method is mainly used for the verification of single speech feature experiment. The specific process is as follows: the target distance between the training samples and the reference speech is calculated by using the training samples in the language library, and then, the parameters of the quadratic fitting curve are calculated by using the curve fitting method according to the principle of minimum variance. According to the parameters of the quadratic fitting curve, the subjective and objective mapping test is carried out, and the corresponding relationship between the objective distortion measurementDk and the subjective estimation MOS is obtained.(6)M=a·D2k+b·Dk+C1<k≤K,wherea, b, C are the parameters of the quadratic fitting curve obtained by training, M is the target distance value, Dk is the objective estimate of MOS subjective score obtained by the quadratic fitting curve, and K is the total number of distortion conditions.(2)
BP scoring method mainly aims at the overall evaluation system of this paper, that is, using the BP algorithm to calculate the objective score of test speech.


## 4.2. Quality Evaluation Index

In recent years, Chinese English learners have made a great breakthrough in the overall level of pronunciation, especially in the pronunciation of monosyllabics (vowel-consonant), such as vowels in the back and change. This is largely due to the country’s investment in educational resources and the enthusiasm of English learners. Therefore, for most English learners, they can better master the pronunciation of English words. But in real life, the rhythm of sentences is the key to English fluency. Many learners encounter a bottleneck after mastering the pronunciation of a single sound because they find that they can pronounce each sound very standard but still can express English with the strong Chinese flavor. This group includes not only beginners but also many intermediate-level learners, English majors, and even some people who use English for a long time. In a sentence, prosody is a very important factor. Every language has its own prosody.The evaluation of voice quality mainly includes voice tone, voice length, and voice rhythm. In the process of evaluation, phonemes and words are mainly evaluated by intonation. Sentences and paragraphs should not only check the content of the pronunciation itself but also determine the true meaning of the sentence to a large extent. When evaluating the pronunciation quality of sentences and paragraphs, we need to consider the prosody information comprehensively, such as whether the speaker can grasp the key information of the sentence more accurately, whether the sentence is not so important, and whether the length of the voice is appropriate. In other words, when evaluating the voice quality of English sentences, good voice quality not only requires complete, no pronunciation errors but also needs speaking speed, accent, and accuracy. Therefore, this paper uses intonation and prosody to evaluate the quality of speech.From the perspective of linguistics, prosody, also known as rhythm, refers to the organization of phonemes higher than the level of segments. It is a systematic organization that organizes various language units into discourse or related chunks in discourse. The realization of prosody can not only convey linguistic information but also paralinguistic and nonverbal information (including emphasis, emphasis, phrase grouping), as well as emotion, attitude, vocabulary discrimination, and other functions. Therefore, the quality of English pronunciation is closely related to its rhythm.Prosodic features, also known as suprasegmental features, mainly refer to speed, rhythm, and intonation composed of dynamic modes such as pitch, intensity, and duration. This paper uses speech speed evaluation and paired variation basic frequency intonation to evaluate and guide English speech quality.


## 4.3. Evaluation Model of Multiparameter Pronunciation Quality

Different groups (such as college students and business people) have different requirements for learning oral English. Based on the evaluation of voice tone, speed, rhythm, intonation, and other quality indicators achieved by taking oral English of college students as the research object, a comprehensive analysis of various indicators is made. In the comprehensive quality evaluation, the relationship between the weight of each index should be considered, and a multiparameter evaluation model and method of college students’ English pronunciation quality should be established to evaluate the pronunciation quality reasonably and objectively.


## 4.4. Speech Evaluation Experiment

The purpose of the speech evaluation experiment is to verify the performance of the English speech quality evaluation model and method proposed in this paper. In this paper, consistency rate and PCC are used to express the manual evaluation.The consistency ratio refers to the ratio of sample size, which is consistent with machine evaluation and nonautomatic evaluation. The specific calculation method is as follows:(7)ASame rate=Evaluate the same number of samplesTotal sample size.The adjacent consistency rate is the ratio between the manual evaluation and the sum of adjacent samples and total samples, in which “adjacent” is defined as one level difference between machine evaluation and manual evaluation. The specific calculation method is as follows:(8)AAdjacent consistency rate=Same number of samples+adjacent samplesTotal sample size.The two variables, which is expressed byr, are the linear correlation between two variables, and the value range is −1∼+1. The greater the absolute value, the stronger the correlation. Generally, 0<R<0.2 indicates very weak correlation, 0.2<R<0.4 indicates very weak correlation, 0.4<R<0.6 indicates medium correlation, 0.6<R<0.8 indicates very strong correlation, and 0.8<R<1 indicates very strong correlation. The specific calculation method is as follows:(9)r=N∑xiyi−∑xi∑yiN∑xi2−∑xi2N∑yi2−∑yi2.


## 5. Experimental Simulation Results and Analysis

### 5.1. Reliability Analysis of Human-Computer Evaluation of Pronunciation Quality

It can be seen from Figure7 that the deviation of a sample number of manual rating and machine rating of the test set in each score segment is kept in a reasonable range. As a whole, this distribution pattern of rating sample conforms to the performance distribution characteristics of general tests, and from the data distribution and data amount, the machine rating meets the general requirements of test rating. However, from the micro point of view, manual rating and machine rating still show differences. The score of manual rating is relatively high, and the machine rating will be “stricter.”Figure 7
Distribution chart of human-machine rating.


### 5.2. Analysis of Speech Recognition Experiment Results

Compared with that of other models, this paper uses the Spanish Arabic digital, which includes 8800 Arabic digital voice data (88 people pronounce 10 Arabic digital voice, and each digit repeats 10 times). 6600 pronunciations of the first 66 subjects were used as training sets, and 2200 pronunciations of the last 22 subjects were used as test sets. The experimental software is matlab82013a.For the same UCI machine learning knowledge base, Arabic spoken digital data set, Nacereddine Hammami, a new model of pattern tree distribution (TDA–GTS) based on tree structure and a new model of pattern tree distribution (TDA–MWST) based on regenerating tree structure are proposed. Compared with the traditional CDHMM and the CDHMM, the recognition effect is improved. Huang Wentao proposed a minimum KASWT weight value, and compared with the BP_Adaboost algorithm, its recognition effect is significantly improved.As can be seen from Figure8, the recognition rate of the DBN model is 97.32%, which is better than the above model. Therefore, the DBN established in this paper is reasonable and effective, which can be further used for speech quality evaluation.Figure 8
Comparison of recognition rates of different methods.


### 5.3. Artificial Evaluation Experiment

In order to facilitate the calculation, according to the method described, after obtaining the scores of 240 statements out of 10 sentences of 24 students, the scores are compared with manual evaluation as shown in Table1 and Figure 9.Table 1
Sample number of experimental results of evaluation index.
Index/sample numberIdenticalDifference grade 1Difference grade 2Difference grade 3Intonation2143520Speech rate1893610Rhythm2113220Intonation1873330Figure 9
Evaluation index experimental results, statistical index.A total of 210 samples were obtained, including 34 first-level samples, 2 second-level samples, and 0 third-level samples. The results show that the coincidence rate of machine and artificial pitch is 87.63%, and the adjacent coincidence rate is 99.79%.In the aspect of speech rate evaluation, there are 185 samples of ME and manual evaluation, 48 samples of the first grade not same, and no samples of the second- or third-grade difference; it shows that the consistency rate of machine and manual speech rate evaluation is 83.21%, the adjacent consistency rate is as high as 99.99%, and the Pearson coefficient is 0.527, which shows that the speech rate EM in this paper is credible.In the aspect of result, there are 212 samples in the same level, 31 samples in the first level, 4 samples in the second level, and no difference in the third level. The results show that the consistency rate of machine and artificial RE is 86.89%, the adjacent consistency rate is 98.63%, and the Pearson coefficient is 0.556.In terms of intonation, there are 184 samples with the same level of ME and manual evaluation, 39 samples with the first level are not the same, only 3 samples with the second level are not the same, and no samples with the third level are not the same. It shows that the consistency rate of machine and manual intonation evaluation is 81.36%, and the adjacent consistency rate is as high as 99.12%. The results show that the intonation evaluation method is very important.To sum up, the assessment methods of intonation, rhythm, and intonation adopted in this paper are reliable and can be further applied to the construction of the English speech quality assessment model.


### 5.4. Test Results of Evaluation Model

This paper evaluates 250 sentences out of 11 sentences of 25 students. The experimental results are shown in Figure10. There are 200 samples for machine evaluation and manual evaluation of the same level, 28 samples for level 1, and no difference between level 2 and level 3 (see Table 2) for specific data.Figure 10
Evaluation model of multiparameter pronunciation quality for college students.Table 2
Overall evaluation test results.
Index/sample numberConsistency rate (%)Adjacent consistency rate (%)PearsonIntonation88.981000.833According to Table2, intonation is the most important, followed by rhythm and intonation, and speed is the least important. After obtaining the above experimental results, we communicated with the pronunciation experts to further discuss the reliability of the proposed multiparameter speech quality evaluation model. Experts believe that intonation is the most important index to evaluate the quality of English pronunciation, which requires accurate content, and pronunciation errors mainly show the emotional color of the speaker, enhance the melody of the voice, and make the voice closer to the reality of life.


## 5.1. Reliability Analysis of Human-Computer Evaluation of Pronunciation Quality

It can be seen from Figure7 that the deviation of a sample number of manual rating and machine rating of the test set in each score segment is kept in a reasonable range. As a whole, this distribution pattern of rating sample conforms to the performance distribution characteristics of general tests, and from the data distribution and data amount, the machine rating meets the general requirements of test rating. However, from the micro point of view, manual rating and machine rating still show differences. The score of manual rating is relatively high, and the machine rating will be “stricter.”Figure 7
Distribution chart of human-machine rating.


## 5.2. Analysis of Speech Recognition Experiment Results

Compared with that of other models, this paper uses the Spanish Arabic digital, which includes 8800 Arabic digital voice data (88 people pronounce 10 Arabic digital voice, and each digit repeats 10 times). 6600 pronunciations of the first 66 subjects were used as training sets, and 2200 pronunciations of the last 22 subjects were used as test sets. The experimental software is matlab82013a.For the same UCI machine learning knowledge base, Arabic spoken digital data set, Nacereddine Hammami, a new model of pattern tree distribution (TDA–GTS) based on tree structure and a new model of pattern tree distribution (TDA–MWST) based on regenerating tree structure are proposed. Compared with the traditional CDHMM and the CDHMM, the recognition effect is improved. Huang Wentao proposed a minimum KASWT weight value, and compared with the BP_Adaboost algorithm, its recognition effect is significantly improved.As can be seen from Figure8, the recognition rate of the DBN model is 97.32%, which is better than the above model. Therefore, the DBN established in this paper is reasonable and effective, which can be further used for speech quality evaluation.Figure 8
Comparison of recognition rates of different methods.


## 5.3. Artificial Evaluation Experiment

In order to facilitate the calculation, according to the method described, after obtaining the scores of 240 statements out of 10 sentences of 24 students, the scores are compared with manual evaluation as shown in Table1 and Figure 9.Table 1
Sample number of experimental results of evaluation index.
Index/sample numberIdenticalDifference grade 1Difference grade 2Difference grade 3Intonation2143520Speech rate1893610Rhythm2113220Intonation1873330Figure 9
Evaluation index experimental results, statistical index.A total of 210 samples were obtained, including 34 first-level samples, 2 second-level samples, and 0 third-level samples. The results show that the coincidence rate of machine and artificial pitch is 87.63%, and the adjacent coincidence rate is 99.79%.In the aspect of speech rate evaluation, there are 185 samples of ME and manual evaluation, 48 samples of the first grade not same, and no samples of the second- or third-grade difference; it shows that the consistency rate of machine and manual speech rate evaluation is 83.21%, the adjacent consistency rate is as high as 99.99%, and the Pearson coefficient is 0.527, which shows that the speech rate EM in this paper is credible.In the aspect of result, there are 212 samples in the same level, 31 samples in the first level, 4 samples in the second level, and no difference in the third level. The results show that the consistency rate of machine and artificial RE is 86.89%, the adjacent consistency rate is 98.63%, and the Pearson coefficient is 0.556.In terms of intonation, there are 184 samples with the same level of ME and manual evaluation, 39 samples with the first level are not the same, only 3 samples with the second level are not the same, and no samples with the third level are not the same. It shows that the consistency rate of machine and manual intonation evaluation is 81.36%, and the adjacent consistency rate is as high as 99.12%. The results show that the intonation evaluation method is very important.To sum up, the assessment methods of intonation, rhythm, and intonation adopted in this paper are reliable and can be further applied to the construction of the English speech quality assessment model.


## 5.4. Test Results of Evaluation Model

This paper evaluates 250 sentences out of 11 sentences of 25 students. The experimental results are shown in Figure10. There are 200 samples for machine evaluation and manual evaluation of the same level, 28 samples for level 1, and no difference between level 2 and level 3 (see Table 2) for specific data.Figure 10
Evaluation model of multiparameter pronunciation quality for college students.Table 2
Overall evaluation test results.
Index/sample numberConsistency rate (%)Adjacent consistency rate (%)PearsonIntonation88.981000.833According to Table2, intonation is the most important, followed by rhythm and intonation, and speed is the least important. After obtaining the above experimental results, we communicated with the pronunciation experts to further discuss the reliability of the proposed multiparameter speech quality evaluation model. Experts believe that intonation is the most important index to evaluate the quality of English pronunciation, which requires accurate content, and pronunciation errors mainly show the emotional color of the speaker, enhance the melody of the voice, and make the voice closer to the reality of life.


## 6. Conclusion

Based on the globalization of China’s economy, English has become a popular language in China. However, due to the Chinese accent, English is regarded as “Chinglish.” However, the traditional speech recognition algorithms DTW, HMM, and ANN are not comprehensive enough, because these speech recognition models ignore that speech quality will be damaged during transmission. Because most of the computer-aided language learning systems at home and abroad focus on the accumulation of grammar and words, the importance of pronunciation is ignored, lack of evaluation indicators in pronunciation as the basis for evaluation, so as to better correct students’ pronunciation and improve oral skills. In addition, the traditional oral test has a strong subjective will, which is only based on manual scoring, so it cannot be fair. In view of the various problems, this paper strengthens the traditional speech recognition technology to consider many factors such as intonation, intonation, speed, and rhythm and further refines the defects of traditional speech recognition technology. From the tables in this paper, it can be clearly concluded that multiparameter pronunciation quality evaluation can be considered, which can give students accurate, effective, and objective evaluation and guidance, improve learners’ oral English learning effect, and achieve a qualitative leap [25].



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*Source: 7846877-2022-04-22.xml*