Control of Electronic Devices Using Neural Network Based Sundanese Speech Recognition

Audio forensics is the application of science and scientific methods in handling digital evidence in the form of audio. In this regard, the audio supports the disclosure of various criminal cases and reveals the necessary information needed in the trial process. So far, research related to audio forensics is more on human voices that are recorded directly, either by using a voice recorder or voice recordings on smartphones, which are available on Google Play services or iOS Store. This study compares the analysis of live voices (human voices) with artificial voices on Google Voice and other artificial voices. This study implements the audio forensic analysis, which involves pitch, formant, and spectrogram as the parameters. Besides, it also analyses the data by using feature extraction using the Mel Frequency Cepstral Coefficient (MFCC) method, the Dynamic Time Warping (DTW) method, and applying the K-Nearest Neighbor (KNN) algorithm. The previously made live voice recording and artificial voice are then cut into words. Then, it tests the chunk from the voice recording. The testing of audio forensic techniques with the Praat application obtained similar words between live and artificial voices and provided 40,74% accuracy of information. While the testing by using the MFCC, DTW, KNN methods with the built systems by using Matlab, obtained similar word information between live voice and artificial voice with an accuracy of 33.33%.

Speech recognition can be defined as the process of converting voice signals into the ranks of the word, by applying a specific algorithm that is implemented in a computer program. The research of speech recognition in Indonesia is relatively limited. This paper has studied methods of feature extraction which is the best among the Linear Predictive Coding (LPC) and Mel Frequency Cepstral Coefficients (MFCC) for speech recognition in Indonesian language. This is important because the method can produce a high accuracy for a particular language does not necessarily produce the same accuracy for other languages, considering every language has different characteristics. Thus this research hopefully can help further accelerate the use of automatic speech recognition for Indonesian language. There are two main processes in speech recognition, feature extraction and recognition. The method used for comparison feature extraction in this study is the LPC and MFCC, while the method of recognition using Hidden Markov Model (HMM). The test results showed that the LPC method is better than MFCC in Indonesian language speech recognition.

The more frequent human interaction to technology demands the development of methods of interaction with the machine to a more natural. Sound which is the most frequently used communication humans makes it one of the natural methods of interaction. Thus the development of a system that can recognize human speech as an action on the machine can be an option for those problems. Voice command is a speech recognition system to give the functions and actions on a system that has been defined as Command and Control systems. Amplitude value is taken from the input sound signal, so we get a set of real numbers is the value input for feature extraction. Feature extraction methods used in this paper are Mel frequency cepstral coefficient (MFCC). Early stages of MFCC is splitting the input signal amplitude values into frames which are processed using the mel-filterbak adapted from the human hearing system. The results of feature extraction made into a codebook which is used as an input symbol on HMM to form a model of every word. On testing step the characteristics of the test signal that has been quantized, then matched with the model that has been created in the previous step. From the test results, the system can recognize spoken words of speakers with an average accuracy of 93.89% in the environment without noise, and 58.1% in environments with noise.

An auditory-based feature extraction algorithm is proposed for enhancing the robustness of automatic speech recognition. In the proposed approach, the speech signal is characterized using a new feature referred to as the Basilar-membrane Frequency-band Cepstral Coefficient (BFCC). In contrast to the conventional Mel-Frequency Cepstral Coefficient (MFCC) method based on a Fourier spectrogram, the proposed BFCC method uses an auditory spectrogram based on a gammachirp wavelet transform in order to more accurately mimic the auditory response of the human ear and improve the noise immunity. In addition, a Hidden Markov Model (HMM) is used for both training and testing purposes. The evaluation results obtained using the AURORA 2 noisy speech database show that compared to the MFCC method, the proposed scheme improves the speech recognition rate by 15% on average given speech samples with Siganl-to-Noise Ratios (SNRs) ranging from 0 to 20 dB. Thus, the proposed method has significant potential for the development of robust speech recognition systems for ambient assisted living.

Voice recognition, command and control systems have played a vital role to facilitate our modern lifestyle and routine operations through hands-free applications. Working people are busy with their routines, simple tasks such as to on/off electrical appliances at home are often overlooked leads to wastage of energy. Besides, the elderly and handicapped often face difficulties to control the electrical appliances due to their physical disability. To overcome, this project aims to develop a voice command intelligent system (VCIS) as an aid to these routine tasks such as to switch on/off a lamp, control a motor and control other modes of operation of home appliances using voice interface. In particular, VCIS involves several development processes such as database collection, pre-processing, feature extraction algorithms and classification modeling for application in smart home. This paper proposes three hierarchical stages in VCIS. The experimental results show that Mel-frequency cepstral coefficients (MFCC) surpassed linear prediction coefficients (LPC) to correlate with the voice commands in all classification stages using artificial neural network. The classification rates were successfully accomplished, 99.12% (MFCC) and 95.23% (LPC). In conclusion, the proposed VCIS can be employed as an efficient method for better quality of life.

Indonesia has so many local cultures and languages that spread from Sabang to Merauke, and most Indonesians do not know how to speak two or more regional languages in Indonesia. Most Indonesians only know familiar tribes that are only commonly heard. Because of this problem, governments in the Indonesian provinces have included lessons on Indonesian culture. The lesson will explain Indonesia’s national and local culture for their area. In this way, the government can also help Indonesians learn more about their local language. This paper proposes a solution to create a learning portal website for users, both from Indonesia and other countries, to learn Indonesian culture and regional languages. In this way, the author hopes that users can learn about Indonesian local culture and language and contribute to preserving Indonesian local culture and language. This application was built on a web-based basis by applying the following HTML, CSS, and javascript using Personal Home Pages (PHP) for database connection using MySQL. The built application was designed using a use case diagram and class diagram to describe the database relations.

Feature extraction techniques for input representation to diagnose infant diseases have received significant attention recently. Mel Frequency Cepstral Coefficients (MFCC) is one of the most popular feature extraction techniques due to its representation method being very similar to the human auditory system. The MFCC method for feature extraction depends on several important parameter settings, namely the number of filter banks, and the number of coefficients used in the final representation. These settings affects the way the features are represented, and in turn, affects the performance of the classifier for diagnosis of the disease. In this paper, the Particle Swarm Optimization (PSO) algorithm was used to optimise the parameters of the MFCC feature extraction method for classifying infants with asphyxia. The extracted MFCC features were then used to train several MLP classifiers over different initialization values. The accuracy of these classifiers was then used to guide the PSO optimization. Our results show that the optimization of MFCC computation using PSO yielded 93.9% accuracy, an improvement of 1.45% over typical MFCC parameter settings using the same classifier.

Indonesian Language has two positions, namely as the national language and the language of the State. In his position as the national language, Indonesian Language serves as a symbol of national pride, a symbol of national identity and as a means of communication between peoples, inter-region and intertribe. Meanwhile, as the language of the state serves as the language of instruction in educational institutions, official state language, national cultureof developer tools, science and technology. The Political Language Seminar in 2000 decided that the position of regional languages serve as a symbol of regional pride, regional identity and local community relations. The presence of Indonesian and regional languages at the present time has begun to erode in the presence of a foreign language. At least, the language users are no longer using language appropriate to the place and the circumstances. Sometimes in a formal situation, people are using foreign languages or regional languages, although the situation is demanding to use Indonesian. The same thing happened in the context of education in Banten province, the vast majority of students and educators in the teaching and learning process in the classroom often use or mix the language with the local language in Indonesian or foreign language. One of the factors that caused the decline in the vitality of Indonesian Language in society and in educational institutions is the linguistic competence of the speakers, thus when doing communication, they mix or even leave Indonesian Language.

Human speech signal is rich with information such as the identity of the speaker, the spoken message, emotional and physical state of the speaker, the spoken language, gender, age and other information. Automatic Speech Recognition (ASR) involves complex tasks aimed at recognition and translation of human speech to text by computers. Phoneme Recognition means recognizing the phonemes associated with speech utterance and is a part of ASR. Developing a phonetic engine and enhancing its performance can lead to significant improvement in ASR. In this paper we propose Artifical Neural Network (ANN) based Phoneme Modeling. We compare performance of speech features such as Inner Hair Cell Coefficients (IHCC) and Mel-Frequency Cepstral Coefficient (MFCC) using various neural network architectures, different optimization algorithms, change in input data vector dimensionality (corresponding to different contextual information) and also increasing the number of epochs. Experiments were carried out on TIMIT database. Our experimental results indicates MFCC performs much better than IHCC feature and the best optimization algorithm is found to be SGD for IHCC and Adagrad for MFCC features by trying out with different sizes of input vector, changing the number of training iterations, training algorithms for various neural network architectures.

Arabic language can be used by native and non- native speakers; due to Arabic is the language of the holy book of Muslims. In this paper, Arabic phoneme recognition system is proposed based on Malay speakers. This system consists of three main stages. The first stage is noise reduction and it aims to enhance the phoneme signals by excluding the unvoiced signals and keep only the voiced signal. Wiener filter is adapted to accomplish this task. The second stage is based on Mel-Frequency Cepstral Coefficients method to extract a vector of features to represent each phoneme signal. Eventually, pattern recognition neural network is designed as recognizer. The proposed system produces sufficient outcomes with 20 hidden neurons. Keywords—Arabic; Malay; pattern recognition; Wiener; Mel- Frequency Cepstral Coefficients I. INTRODUCTION It is well known the wide range of Artificial Neural Networks (ANN) application in speech recognition, financial, telecommunications, electronics and medical. As for speech recognition system for Arabic language, (1) & (2) have addressed the implementation of recognition system based on Arabic native speakers. In general, neural networks (NNs) for phonemes recognition are divided into several categories for instance Very Large Scale Integration (VLSI) NN that can be divided into digital and analogue with digital NNs being more compatible with feed-forward neural networks while analogue NNs are found to be more successful with recurrent NNs (3). From Automatic Speech System (ASR) point of view, NNs for recognition purpose can be divided into two main category namely conventional neural networks such as Multilayer Perceptron (MLP) and Radial Basis Function (RBF) and secondly Recurrent Neural Networks (RNN). The first category of NNs was implemented as pattern classifiers and proven capable for recognition purpose but not at par as compared to Hidden Markov Model (HMM) performance (2). At present, HMM has been proven to be the most successful approach in ASR research area until recently the possibility to hybrid both HMM and ANN (4) & (5). In general, speech recognition systems consist of three stages specifically noise reduction, feature extraction and classification. The aim of noise reduction is to reduce the level of noise in the speech signal and make it more compatible for the ASR. As for feature extraction, this process attempted to extract the most relevant features related to the phoneme signal. The extracted features acted as input features of the NN during training and testing. Finally, the purpose of classification process is to arrange the most similar and related features in categories based on the training phase. Basically, networks training database are being observed and updated during training session of the networks, involving the weights and biases arguments. Networks with lower errors can be considered as better networks (6). Hence, this paper deems to explore further the ability of NN for Arabic phonemes recognition spoken by Malay individuals. This paper is arranged as following; related work is discussed in section II followed by section III that discussed method to be implemented based on Wiener filter, Mel-Frequency Cepstral Coefficients and pattern recognition neural network. Section IV elaborated experimental analysis and results and followed by Conclusion section.

Telugu is the standard language used to communicate mainly in Andhra Pradesh and Telangana states with approximately 100 million speakers. Every Telugu word ends with vowels. Automatic speech recognition has major applications in the smart world. Telugu speech recognition has a huge scope of research from past decade. This paper deals with Telugu speech recognition in a speaker-dependent format for 10 words which are numbers from one to ten uttered by 10 speakers which creates a data base of 100 samples. Numbers are very frequently spoken words where recognition of these words plays a major role in Telugu speech recognition. These spoken words are preprocessed using various techniques like de-noising, framing, sampling, transformations, and endpoint detection. Methods like mel frequency cepstral coefficients (MFCC) and combined MFCC with modified group delay functions (MODGDF) are used for extracting the features. Extracted features are used for training and testing phase. Multilayer perceptron (MLP) and time lagged recurrent neural network (TLRN) patterns are trained and tested. Comparison is done for MLP and TLRN with the feature extraction techniques MFCC and MODGDF-MFCC. Integrated MODGDF-MFCC gives best accuracy in training and testing phases.

The acoustic classification technology of moving vehicles is a significant application in wireless sensor networks (WSNs). However, the wild environment makes it more intractable for the acoustic target classification owing to the complicated interference noise. The widely used mel-frequency cepstral coefficients (MFCCs) are somewhile contaminated by the acoustic noise, resulting in degrading the classification performance in real environments. To increase the classification accuracy of the moving vehicles, this paper presents an acoustic feature extraction method, which integrates the nonlinear function with the MFCC method, termed as NMFCC. Through nonlinear transformation, the proposed method could amplify the acoustic features in a nonlinear way, and obtain more robust acoustic features, thereby achieving a superior classification performance. Comparing the NMFCC with the baseline feature extraction method, experimental results not only demonstrate the viability of the proposed algorithm but also show a satisfactory performance of the moving vehicle classification for diverse practical scenarios in the wild.

Optimisation of phonetic aware speech recognition through multi-objective evolutionary algorithms

While there has been substantial amount of work in speaker diarization recently, there are few efforts in jointly employing lexical and acoustic information for speaker segmentation. Towards that, we investigate a speaker diarization system using a sequence-to-sequence neural network trained on both lexical and acoustic features. We also propose a loss function that allows for selecting not only the speaker change points but also the best speaker at any time by allowing for different speaker groupings. We incorporate Mel Frequency Cepstral Coefficients (MFCC) as an acoustic feature alongside lexical information that are obtained from conversations from the Fisher dataset. Thus, we show that acoustics provide complementary information to the lexical modality. The experimental results show that sequence-to-sequence system trained on both word sequences and MFCC can improve on speaker diarization result compared to the system that only relies on lexical modality or the baseline MFCC-based system. In addition, we test the performance of our proposed method with Automatic Speech Recognition (ASR) transcripts. While the performance on ASR transcripts drops, the Diarization Error Rate (DER) of our proposed method still outperforms the traditional method based on Bayesian Information Criterion (BIC).

Over the past few decades, research has focused heavily on automatic speech recognition (ASR). Although ASR for a few languages is close to reality; ASR for the low resource languages like Malayalam is still in its infancy. Here, in this work, the authors discuss the experiment conducted on accented data of Malayalam speech using two approaches. One approach uses the spectral features for modeling using deep convolutional neural network and the other uses the influential features of speech signals for modeling using LSTM-RNN approach. The proposed methodology comprises three distinct stages; dataset preparation, feature extraction, classification, and hence the construction of deep learning models that recognize the accent-based spoken sentences in the Malayalam language. Mel-frequency cepstral coefficient (MFCC) algorithm, short-term Fourier transform (STFT), and mel spectrogram methodologies are used for feature engineering and hence the features that represent the speech signals are used for constructing the accented ASR system for the Malayalam language using long short-term memory (LSTM) a recurrent neural network (RNN). The spectrogram dataset has been constructed for the speech dataset and used for constructing the ASR model with deep convolutional neural network (DCNN). The result of the experiment shows that LSTM-based RNN outperforms DCNN for the proposed dataset that has been constructed for the experiment in the natural recording environment.