Mel Frequency Cepstral Coefficients Features Research Articles

Enhancing Speech Emotion Recognition Using Dual Feature Extraction Encoders.

Understanding and identifying emotional cues in human speech is a crucial aspect of human-computer communication. The application of computer technology in dissecting and deciphering emotions, along with the extraction of relevant emotional characteristics from speech, forms a significant part of this process. The objective of this study was to architect an innovative framework for speech emotion recognition predicated on spectrograms and semantic feature transcribers, aiming to bolster performance precision by acknowledging the conspicuous inadequacies in extant methodologies and rectifying them. To procure invaluable attributes for speech detection, this investigation leveraged two divergent strategies. Primarily, a wholly convolutional neural network model was engaged to transcribe speech spectrograms. Subsequently, a cutting-edge Mel-frequency cepstral coefficient feature abstraction approach was adopted and integrated with Speech2Vec for semantic feature encoding. These dual forms of attributes underwent individual processing before they were channeled into a long short-term memory network and a comprehensive connected layer for supplementary representation. By doing so, we aimed to bolster the sophistication and efficacy of our speech emotion detection model, thereby enhancing its potential to accurately recognize and interpret emotion from human speech. The proposed mechanism underwent a rigorous evaluation process employing two distinct databases: RAVDESS and EMO-DB. The outcome displayed a predominant performance when juxtaposed with established models, registering an impressive accuracy of 94.8% on the RAVDESS dataset and a commendable 94.0% on the EMO-DB dataset. This superior performance underscores the efficacy of our innovative system in the realm of speech emotion recognition, as it outperforms current frameworks in accuracy metrics.

CovidCoughNet: A new method based on convolutional neural networks and deep feature extraction using pitch-shifting data augmentation for covid-19 detection from cough, breath, and voice signals

This study proposes a new deep learning-based method that demonstrates high performance in detecting Covid-19 disease from cough, breath, and voice signals. This impressive method, named CovidCoughNet, consists of a deep feature extraction network (InceptionFireNet) and a prediction network (DeepConvNet). The InceptionFireNet architecture, based on Inception and Fire modules, was designed to extract important feature maps. The DeepConvNet architecture, which is made up of convolutional neural network blocks, was developed to predict the feature vectors obtained from the InceptionFireNet architecture. The COUGHVID dataset containing cough data and the Coswara dataset containing cough, breath, and voice signals were used as the data sets. The pitch-shifting technique was used to data augmentation the signal data, which significantly contributed to improving performance. Additionally, Chroma features (CF), Root mean square energy (RMSE), Spectral centroid (SC), Spectral bandwidth (SB), Spectral rolloff (SR), Zero crossing rate (ZCR), and Mel frequency cepstral coefficients (MFCC) feature extraction techniques were used to extract important features from voice signals. Experimental studies have shown that using the pitch-shifting technique improved performance by around 3% compared to raw signals. When the proposed model was used with the COUGHVID dataset (Healthy, Covid-19, and Symptomatic), a high performance of 99.19% accuracy, 0.99 precision, 0.98 recall, 0.98 F1-Score, 97.77% specificity, and 98.44% AUC was achieved. Similarly, when the voice data in the Coswara dataset was used, higher performance was achieved compared to the cough and breath studies, with 99.63% accuracy, 100% precision, 0.99 recall, 0.99 F1-Score, 99.24% specificity, and 99.24% AUC. Moreover, when compared with current studies in the literature, the proposed model was observed to exhibit highly successful performance. The codes and details of the experimental studies can be accessed from the relevant Github page: (https://github.com/GaffariCelik/CovidCoughNet).

Automatic speaker verification system for dysarthric speakers using prosodic features and out-of-domain data augmentation

A communication disorder is an impairment of a person’s ability to talk or communicate appropriately. Dysarthria is a common neuro-motor speech communication disorder that can be caused by neurological damage. Dysarthria may affect the articulation, phonation, and prosody of a speaker. Dysarthria patients have poor neuromotor coordination and other physical impairments, making it difficult to utilize an interactive keyboard or other user interfaces. The ASV system can make biometric applications more accessible to dysarthric speakers by eliminating the need for them to remember cumbersome and unique authentication numbers and passwords. In this paper, we presented a study on developing an automatic speaker verification (ASV) system for dysarthria patients with varying speech intelligibility to assist them in remote access control and voice-based biometric applications. In the initial part of our proposed approach, we included a duration modification-based data augmentation module in the front end of the ASV system. Since prosody deficits are one of the early indicators of dysarthria, we investigated the role of prosodic variables in combination with the traditional Mel-frequency cepstral coefficients (MFCC). The prosodic variables explored in this study include pitch, loudness, and voicing probability. Separate i-vector and x-vector models are trained and compared using individual MFCC, prosodic variables, and their combinations. The experimental results showed that the proposed approach based on combining MFCC and prosody features along with duration-modification-based data augmentation produced promising results.

Recognizing Command Words using Deep Recurrent Neural Network for Both Acoustic and Throat Speech

The importance of speech command recognition in a human-machine interaction system is increased in recent years. In this study, we propose a deep neural network-based system for acoustic and throat command speech recognition. We apply a preprocessed pipeline to create the input of the deep learning model. Firstly, speech commands are decomposed into components using well-known signal decomposition techniques. The Mel-frequency cepstral coefficients (MFCC) feature extraction method is applied to each component of the speech commands to obtain the feature inputs for the recognition system. At this stage, we apply and compare performance using different speech decomposition techniques such as wavelet packet decomposition (WPD), continuous wavelet transform (CWT), and empirical mode decomposition (EMD) in order to find out the best technique for our model. We observe that WPD shows the best performance in terms of classification accuracy. This paper investigates long short-term memory (LSTM)-based recurrent neural network (RNN), which is trained using the extracted MFCC features. The proposed neural network is trained and tested using acoustic speech commands. Moreover, we also train and test the proposed model using a throat mic. speech commands as well. Lastly, the transfer learning technique is employed to increase the test accuracy for throat speech recognition. The weights of the model train with the acoustic signal are used to initialize the model used for throat speech recognition. Overall, we have found significant classification accuracy for both acoustic and throat command speech. We obtain LSTM is much better than the GMM-HMM model, convolutional neural networks such as CNN-tpool2 and residual networks such as res15 and res26 with an accuracy score of over 97% on Google’s Speech Commands dataset and we achieve 95.35% accuracy on our throat speech data set using the transfer learning technique.

SWMAT: Mel-frequency cepstral coefficients-based memory fingerprinting for IoT devices

The increasing sophistication in computing capability and sensing technologies have continued to drive the design, development, and growth of the smart technologies commonly known as the IoTs. Nonetheless, the rise and spread of malware in this ecosystem is a pressing societal concern that requires immediate attention. In this paper, we propose a novel technique called Sound Wave Memory Analysis Technique (SWMAT), for fingerprinting IoT devices by converting their dynamic memory traces into sound wave signals using a lossless transformation function from which a unique set of determinable features called Mel Frequency Cepstral Coefficients (MFCCs) are extracted. The overarching objective of this research is to explore offline the effectiveness of using features from memory-encoded sound wave signals for fingerprinting and detecting abnormal changes in IoT devices, which potentially can provide an excellent technique for an on-device Host-Based Intrusion Detection System. Our SWMAT scores the similarity between two sequences of MFCCs using a Dynamic Time Warping distance measure. To evaluate our approach, we developed multiple IoT testbeds and generated 125 MFCC sequences from 20 benign and 5 infected IoT applications. Our results showed that the MFCC features, when leveraged as fingerprints for both Intra and Inter-app similarity, can uniquely distinguish an IoT process and can detect when an IoT process has been hijacked and/or is modified by another malicious code. Furthermore, this empirical result shows our technique’s similarity detection accuracy to be ≈95%.

Effective preservation of higher-frequency contents in the context of short utterance based children’s speaker verification system

Developing an automatic speaker verification (ASV) system for children is extremely challenging due to the unavailability of children’s speech corpora. The challenges are further exacerbated in the case of short utterances. Voice-based biometric systems require adequate amount of speech data for enrollment and verification; otherwise the performance considerably degrades. In this paper, we have focussed on data paucity and preserving the higher-frequency contents in order to enhance the performance of a short-utterance based children’s speaker verification system. To deal with data scarcity, several out-of-domain data augmentation techniques have been utilized. Since the out-of-domain data used is from adult speakers which are acoustically very different from children’s speech, we have resorted to techniques like prosody modification, formant modification and voice-conversion in order to render it acoustically similar to children’s speech prior to augmentation. This helps in not only increasing the amount of training data but also in effectively capturing the missing target attributes. A relative improvement of 33.6% in equal error rate (EER) with respect to the baseline system trained solely on child data-set is achieved when the proposed data augmentation technique is applied. Further to that, for the preservation of the higher-frequency contents, we have resorted to concatenation of the classical Mel-frequency cepstral coefficients (MFCC) features with the linear-frequency cepstral coefficient (LFCC) or with the inverse-Mel-frequency cepstral coefficient (IMFCC) features. The use of Mel-filter-bank leads to poor resolution of higher-frequency components. On the other hand, linear- or inverse-Mel-filter-banks yield better resolution of higher-frequency components. Moreover, MFCC and IMFCC features exhibit low canonical correlation. Consequently, the frame-level concatenation of MFCC and LFCC or IMFCC features leads to better resolution of both lower- as well as higher-frequency components. Therefore, the EER considerably reduces when either LFCC features or IMFCC features are concatenated with MFCC features. The EER for the full test set shows a relative reduction of 10.56% (with respect to the EER for the MFCC features) when IMFCC features are concatenated with the MFCC features. This novel approach of incorporating data augmentation followed by frame-level feature concatenation helps in achieving an overall reduction of 40.6% in EER.

Time-frequency analysis of speech signal using Chirplet transform for automatic diagnosis of Parkinson's disease.

Parkinson's disease (PD) is the second most prevalent neurodegenerative disorder in the world after Alzheimer's disease. Early diagnosing PD is challenging as it evolved slowly, and its symptoms eventuate gradually. Recent studies have demonstrated that changes in speech may be utilized as an excellent biomarker for the early diagnosis of PD. In this study, we have proposed a Chirplet transform (CT) based novel approach for diagnosing PD using speech signals. We employed CT to get the time-frequency matrix (TFM) of each speech recording, and we extracted time-frequency based entropy (TFE) features from the TFM. The statistical analysis demonstrates that the TFE features reflect the changes in speech that occurs in the speech due to PD, hence can be used for classifying the PD and healthy control (HC) individuals. The effectiveness of the proposed framework is validated using the vowels and words from the PC-GITA database. The genetic algorithm is utilized to select the optimum features subset, while a support vector machine (SVM), decision tree (DT), K-Nearest Neighbor (KNN), and Naïve Bayes (NB) classifiers are employed for classification. The TFE features outperform the breathiness and Mel frequency cepstral coefficients (MFCC) features. The SVM classifier is most effective compared to other machine-learning classifiers. The highest classification accuracy rates of 98% and 99% are achieved using the vowel /a/ and word /atleta/, respectively. The results reveal that the proposed CT-based entropy features effectively diagnose PD using the speech of a person.

Multichannel CNN-BLSTM Architecture for Speech Emotion Recognition System by Fusion of Magnitude and Phase Spectral Features Using DCCA for Consumer Applications

Conventional Speech Emotion Recognition (SER) approaches put more emphasis on extracting magnitude spectrum-based features, such as Mel Frequency Cepstral Coefficients (MFCCs), and Mel spectrogram. However, phase information is ignored due to signal processing difficulties such as the phase wrapping issue. This work develops a multichannel Convolution Neural Network-Bidirectional Long Short Term Memory (CNN-BLSTM) architectures with an attention mechanism for speaker-independent SER by considering phase and magnitude spectrum-based features. The phase-based features are extracted using the Modified Group Delay Function (MODGD). The obtained phase features are combined with MFCC features. The CNN-BLSTM network extract learned representation from magnitude and phase features. The learned representation from MFCCs and MODGD are combined and given as an input to the Support Vector Machine (SVM) for classification. The Deep Canonical Correlation Analysis (DCCA) is used to maximize the correlation between magnitude and phase information to improve the conventional SER system’s performance. Here the IEMOCAP database is used for performance analysis. The experimental results show improvement over MFCC features and existing approaches for unimodal SER. In this work, we also developed real-time web server application for the proposed architecture.

Tuning Dari Speech Classification Employing Deep Neural Networks

Recently, many researchers have focused on building and improving speech recognition systems to facilitate and enhance human-computer interaction. Today, Automatic Speech Recognition (ASR) system has become an important and common tool from games to translation systems, robots, and so on. However, there is still a need for research on speech recognition systems for low-resource languages. This article deals with the recognition of a separate word for Dari language, using Mel-frequency cepstral coefficients (MFCCs) feature extraction method and three different deep neural networks including Convolutional Neural Network (CNN), Recurrent Neural Network (RNN), Multilayer Perceptron (MLP). We evaluate our models on our built-in isolated Dari words corpus that consists of 1000 utterances for 20 short Dari terms. This study obtained the impressive result of 98.365% average accuracy.

Interfacial debonding detection of steel beams reinforced by CFRP plates based on percussion method

Steel structure reinforced engineering makes extensive use of carbon fiber reinforced plastic (CFRP) plates. Although it happens frequently, the interfacial debonding of CFRP plate-reinforced steel beams has been the main cause of failure in flexural strengthened steel beams, which results in the brittle failure of steel structures. This paper proposes a novel percussion method to locate and quantify the debonding area based on the Welch power spectrum density (PSD) estimate, mel frequency cepstral coefficient (MFCC), and convolutional neural network (CNN). According to our knowledge, this is the first attempt to use the percussion method to identify and measure the debonding defects of steel beams strengthened by CFRP plates. Additionally, this experiment takes into account how percussion instruments affect localization and quantification. To begin with, the debonding location is qualitatively detected using the Welch PSD estimate method. The presented percussion method can successfully and accurately detect the interfacial debonding of CFRP plate-reinforced steel beams, according to repeated results. Subsequently, a novel CNN model is constructed to classify the various debonding areas based on the MFCC feature. In the end, the experimental findings show that the localization and quantification of the hammer types vary only slightly. Overall, the percussion-based approach can offer a fresh line of inquiry into the debonding problem on steel beams reinforced with CFRP plates.

Kiñit classification in Ethiopian chants, Azmaris and modern music: A new dataset and CNN benchmark.

In this paper, we create EMIR, the first-ever Music Information Retrieval dataset for Ethiopian music. EMIR is freely available for research purposes and contains 600 sample recordings of Orthodox Tewahedo chants, traditional Azmari songs and contemporary Ethiopian secular music. Each sample is classified by five expert judges into one of four well-known Ethiopian Kiñits, Tizita, Bati, Ambassel and Anchihoye. Each Kiñit uses its own pentatonic scale and also has its own stylistic characteristics. Thus, Kiñit classification needs to combine scale identification with genre recognition. After describing the dataset, we present the Ethio Kiñits Model (EKM), based on VGG, for classifying the EMIR clips. In Experiment 1, we investigated whether Filterbank, Mel-spectrogram, Chroma, or Mel-frequency Cepstral coefficient (MFCC) features work best for Kiñit classification using EKM. MFCC was found to be superior and was therefore adopted for Experiment 2, where the performance of EKM models using MFCC was compared using three different audio sample lengths. 3s length gave the best results. In Experiment 3, EKM and four existing models were compared on the EMIR dataset: AlexNet, ResNet50, VGG16 and LSTM. EKM was found to have the best accuracy (95.00%) as well as the fastest training time. However, the performance of VGG16 (93.00%) was found not to be significantly worse (P < 0.01). We hope this work will encourage others to explore Ethiopian music and to experiment with other models for Kiñit classification.

Wavelet-based Parametric Feature Subset Selection for Speaker and Accent Recognition using Genetic Algorithm

Research on speaker and accent recognition studies using the Malay language in the field of Automatic Speech Recognition (ASR) is limited, with most studies focusing on speech recognition. This study proposes to increase the performance Malaysian speakers and accent recognition using wavelets transform, namely Wavelet Packet Transform (WPT) and Dual-Tree Complex Wavelet Packet Transform (DT-CWPT). A variety of feature extraction combinations, including conventional Mel Frequency Cepstral Coefficients (MFCC), Linear Predictive Coding (LPC) and wavelets transform, were implemented to compare the effectiveness of the proposed method. Although the proposed approach resulted in improved detection rate, it faced challenges in terms of high feature dimensionality and increased computation time. To address these issues, the Genetic Algorithm (GA) approach has been adopted to reduce the number of irrelevant features, accelerate the learning system and achieve better performance. The extracted features were trained using various classifiers, including k-Nearest Neighbors (k-NN), Support Vector Machine (SVM) and Extreme Learning Machine (ELM). The experimental results showed that the best speaker recognition accuracy was 97.33% for English numbers using SVM classifier and 96.02% for Malay words using the ELM classifier with a combination of wavelets, LPC and MFCC features. For accent recognition, the ELM classifier yielded the best performance, achieving 95.28% accuracy for English numbers with a combination of wavelets and MFCC features and 96.72% for Malay words using combined feature extraction of wavelets, LPC and MFCC feature extraction. It can be concluded that Malay words yielded better recognition rates than English numbers. Furthermore, use of GA effectively reduced the overall number of features while maintaining high accuracy level.

A Perspective Study on Speech Recognition

Emotions play an extremely important role in human mental life. It is a medium of expression of one’s perspective or one’s mental state to others. Speech Emotion Recognition (SER) can be defined as extraction of the emotional state of the speaker from his or her speech signal. There are few universal emotions including Neutral, Anger, Happiness, and Sadness in which any intelligent system with finite computational resources can be trained to identify or synthesize as required. In this work spectral and prosodic features are used for speech emotion recognition because both of these features contain the emotional information. Mel-Frequency Cepstral Coefficients (MFCC) is one of the spectral features. Fundamental frequency, loudness, pitch and speech intensity and glottal parameters are the prosodic features which are used to model different emotions. The potential features are extracted from each utterance for the computational mapping between emotions and speech patterns. Pitch can be detected from the selected features, using which gender can be classified. The audio signal is filtered using a method known as feature extraction technique. In this article, the feature extraction technique for speech recognition and voice classification is analyzed and also centered to comparative analysis of different types of Mel frequency cepstral coefficients (MFCC) feature extraction method. The MFCC technique is used for deduction of noise in voice signals and also used for voice classification and speaker identification. The statistical results of the different MFCC techniques are discussed and finally concluded that the delta-delta MFCC feature extraction technique is better than the other feature extraction techniques..

Tuning Dari Speech Classification Employing Deep Neural Networks

Recently, many researchers have focused on building and improving speech recognition systems to facilitate and enhance human-computer interaction. Today, Automatic Speech Recognition (ASR) system has become an important and common tool from games to translation systems, robots, and so on. However, there is still a need for research on speech recognition systems for low-resource languages. This article deals with the recognition of a separate word for Dari language, using Mel-frequency cepstral coefficients (MFCCs) feature extraction method and three different deep neural networks including Convolutional Neural Network (CNN), Recurrent Neural Network (RNN), Multilayer Perceptron (MLP), and two hybrid models of CNN and RNN. We evaluate our models on our built-in isolated Dari words corpus that consists of 1000 utterances for 20 short Dari terms. This study obtained the impressive result of 98.365% average accuracy.

A late fusion deep neural network for robust speaker identification using raw waveforms and gammatone cepstral coefficients

Speaker identification aims at determining the speaker identity by analyzing his voice characteristics, and relies typically on statistical models or machine learning techniques. Frequency-domain features are by far the most used choice to encode the audio input in sound recognition. Recently, some studies have also analyzed the use of time-domain raw waveform (RW) with deep neural network (DNN) architectures. In this paper, we hypothesize that both time-domain and frequency-domain features can be used to increase the robustness of speaker identification task in adverse noisy and reverberation conditions, and we present a method based on a late fusion DNN using RWs and gammatone cepstral coefficients (GTCCs). We analyze the characteristics of RW and spectrum-based short-time features, reporting advantages and limitations, and we show that the joint use can increase the identification accuracy. The proposed late fusion DNN model consists of two independent DNN branches made primarily by convolutional neural networks (CNN) and fully connected neural networks (NN) layers. The two DNN branches have as input short-time RW audio fragments and GTCCs, respectively. The late fusion is computed on the predicted scores of the DNN branches. Since the method is based on short segments, it has the advantage of being independent from the size of the input audio signal, and the identification task can be computed by summing the predicted scores over several short-time frames. Analysis of speaker identification performance computed with simulations show that the late fusion DNN model improves the accuracy rate in adverse noise and reverberation conditions in comparison to the RW, the GTCC, and the mel-frequency cepstral coefficients (MFCCs) features. Experiments with real-world speech datasets confirm the efficiency of the proposed method, especially with small-size audio samples.

Underwater Acoustic Target Recognition Based on Data Augmentation and Residual CNN

In the field of underwater acoustic recognition, machine learning methods rely on a large number of datasets to achieve high accuracy, while the actual collected signal samples are often very scarce, which has a great impact on the recognition performance. This paper presents a recognition method of an underwater acoustic target by the data augmentation technique and the residual convolutional neural network (CNN) model, which is used to expand training samples to improve recognition performance. As a representative model in residual CNN, the ResNet18 model is used for recognition. The whole process mainly includes mel-frequency cepstral coefficient (MFCC) feature extraction, data augmentation processing, and ResNet18 model recognition. On the base of the traditional data augmentation, this study used the deep convolutional generative adversarial network (DCGAN) model to realize the expansion of underwater acoustic samples and compared the recognition performance of support vector machine (SVM), common CNN, VGG19, and ResNet18. The recognition results of the MFCC, constant Q transform (CQT), and low-frequency analyzer and recorder (LOFAR) spectrum were also analyzed and compared. Experimental results showed that the recognition accuracy of the MFCC feature was better than that of other features at the same method, and using the data augmentation method could obviously improve the recognition performance. Moreover, the recognition performance of ResNet18 using data enhancement technology was better than that of other models, which was due to the combination of the data expansion advantage of data augmentation technology and the deep feature extracting ability of the residual CNN model. In addition, although this method was used for ship recognition in this paper, it is not limited to this. This method is also applicable to other target voice recognition, such as natural sound and underwater voice biometrics.

Closed-set automatic speaker identification using multi-scale recurrent networks in non-native children.

Children may benefit from automatic speaker identification in a variety of applications, including child security, safety, and education. The key focus of this study is to develop a closed-set child speaker identification system for non-native speakers of English in both text-dependent and text-independent speech tasks in order to track how the speaker's fluency affects the system. The multi-scale wavelet scattering transform is used to compensate for concerns like the loss of high-frequency information caused by the most widely used mel frequency cepstral coefficients feature extractor. The proposed large-scale speaker identification system succeeds well by employing wavelet scattered Bi-LSTM. While this procedure is used to identify non-native children in multiple classes, average values of accuracy, precision, recall, and F-measure are being used to assess the performance of the model in text-independent and text-dependent tasks, which outperforms the existing models.

Comparative Analysis of LPC and MFCC for Male Speaker Recognition in Text-Independent Context

peech is the utmost communication medium for human beings which conveys rich and valuable information such as accent, gender, emotion and unique identity. Therefore, automatic speaker recognition can be developed based on unique characteristics of one’s speech and utilized for applications such as voice dialing, online banking, and telephone shopping to verify the identity of its users. However, retrieving salient features which are capable of identifying speakers is a challenging problem in speech recognition systems since speech contains abundant information. In this study, a total of 438 audio data obtained from speakers uttering speech in text-independent context is proposed using speech data elicited from three Malay male speakers. The performance of two popularly used feature extraction techniques namely, linear prediction coefficients (LPC) and Mel-frequency cepstral coefficients (MFCC) were compared using discriminant analysis model. Although both features yielded impressive outcomes, the MFCC features surpassed that of LPC by achieving a higher accuracy rate of 99.09%, which was 4.34% higher than the latter.

SPOKEN-DIGIT CLASSIFICATION USING ARTIFICIAL NEURAL NETWORK

Audio classification has been one of the most popular applications of Artificial Neural Networks. This process is at the center of modern AI technology, such as virtual assistants, automatic speech recognition, and text-to-speech applications. There have been studies about spoken digit classification and its applications. However, to the best of the author's knowledge, very few works focusing on English spoken digit recognition that implemented ANN classification have been done. In this study, the authors utilized the Mel-Frequency Cepstral Coefficients (MFCC) features of the audio recording and Artificial Neural Network (ANN) as the classifier to recognize the spoken digit by the speaker. The Audio MNIST dataset was used as training and test data while the Free-Spoken Digit Dataset was used as additional validation data. The model showed an F-1 score of 99.56% accuracy for the test data and an F1 score of 81.92% accuracy for the validation data.

The design of intelligent fuzzy cognitive system of music emotion by product supply chain management

AbstractThe classification of musical emotion is a challenging problem due to the subjectivity of human perception. In order to analyse the emotion contained in music from the perspective of product supply chain management, the psychological model and characteristics of music emotion are introduced. Additionally, the current product supply chain and the composition of the two contracts are analysed based on the characteristics of online music products. Finally, a speech emotion recognition model based on a Fuzzy Cognitive Map (FCM) is proposed. Two corpora containing five emotions (sadness, anger, happiness, surprise, and boredom) are selected to test the performance and characteristics of the network. In order to evaluate the system performance, different acoustic features are extracted, including prosodic features, formant features, marginal spectral features, and Mel‐Frequency Cepstral Coefficients (MFCC). The results show that the average recognition rate of MFCC features on the Berlin emotional speech database is 70.36%, which is 0.75% higher than the second‐highest edge spectrum features and achieves better results. Meanwhile, on this database, the Musical Emotion Recognition System based on FCM (MERS‐FCM) is improved by 20.36%, 7.34%, and 4.12% compared with Back Propagation, K‐nearest neighbour, and support vector machine, respectively, which shows that MERS‐FCM has better performance. The research of the music emotion recognition system based on FCM can provide more reference paths for the use and development of music emotion information.