Abstract

Designing various learning strategies has been gaining a lot of scientific interest during the recent progress of deep learning methodologies. Curriculum learning is a learning strategy aimed at training the neural network model by presenting the samples in a specific meaningful order rather than randomly sampling the training examples from the data distribution. In this work, we have explored starting small paradigm of curriculum learning technique for speech recognition. The starting small paradigm of curriculum learning is performed by a two step learning strategy. Training dataset is re-organized as a set of easily classifiable examples followed by the actual training dataset and the model is trained on the re-organized dataset. We hypothesize that by following the starting small learning paradigm the learning gets initialized in a better way and progresses to attain a better convergence. We propose to rank the toughness of the training example based on the posterior probabilities obtained using a previously trained model. Apart from re-arranging the training corpus starting small paradigm of curriculum learning is applied at model level. We consider the broad manner-class classification objective function as the smoother version of the phone class classification objective function. The model initially trained for broad class classification is later adapted for phone classification.In this work, we have used TIMIT and a subset of WallStreet Journal (WSJ) corpus to validate the experiments, both the learning strategies have shown consistently better performances across the two datasets compared to the baseline system trained by randomly sampling the dataset.

Abstract

A high degree of robustness is a prerequisite to operate speech and language processing systems in practical environments. Performance of these systems is highly influenced by varying and mixed background environments. In this pa-per, we put forward a robust method for automatic language identification in various background environments. Combined temporal and spectral processing method is used as a preprocessing technique for enhancing the degraded speech. Language discriminative information in high sonority regions of speech is used for the task of language identification. Sonority regions are regions of speech whose signal energy is high and these regions are less influenced by background environments.Spectral energy of formants in the glottal closure regions is employed as an acoustic correlate for the detection of sonority regions of speech. In this paper performance of the LID system is studied in various background environments like clean room, car factory,high frequency,pink and white noise environments. In this work, Indian Institute of Technology Kharagpur - Multi LingualI ndian Language Speech Corpus (IITKGP-MLILSC) is used for building language identification system. Noise speech samples from the NOISEX database are employed in the present study.The performance of the proposed method is quite satisfactory compared to existing approaches.

Abstract

Most of the Indian languages are originated from Devanagari, the script of the Sanskrit language. In-spite of similarity in phoneme sets, every language its own influence on the phonotactic constraints of speech in that language. A modelling technique that is capable of capturing the slightest variations imparted by the language is a prerequisite for developing a language identification system (LID). Use of Gaussian mixture modelling technique with a large number of mixture components demands a large training data for each language class, which is hard to collect and handle. In this work, phonotactic variations imparted by the different languages are modelled using Gaussian mixture modelling with a universal background model (GMM-UBM) technique.In GMM-UBM based modelling certain amount of data from all the language classes is pooled to develop a universal background model (UBM) and the model is adapted to each class. Spectral features (MFCC) are employed to represent the language specific phonotactic information of speech in different languages. During the present study, LID systems are developed using the speech samples from IITKGP-MLILSC. In this work, performance of the proposed GMM-UBM based LID system is compared with conventional GMM based LID system. An average improvement of 7–8% is observed due to the use of UBM-based modelling of developing a LID system.

Abstract

In recent past a lot of scientific attention is paid on recognizing the emotional state of the speaker from his speech. Emotion recognition is a challenging task as human emotions are complex, subtle and emotive state in human speech does not persist long. So it is important to study the presence of emotion identifiable information in smaller segments of speech. This study is aimed at studying the presence of emotional specific information with relevance to the position of the word in the utterance. During the present study, spectral features are employed to represent emotion specific information in speech. Spectral features from smaller speech segments of speech based on their position in the utterance are employed to study the presence of emotion in speech. Due to the lack of adequate data in small speech segments to support conventional GMM during the course of present study Gaussian mixture modeling with a universal background model (GMM-UBM) is used for developing a emotion recognition system. Speech data from IITKGP-SESC is used during the course of the present study. During the present study 4 (Anger, Fear, Happy and Neutral) emotions are considered.

Abstract

Emotions in human speech are short lived. In an emotive utterance, the emotive gestures produced due to the emotive state of the speaker persists only to a shorter duration. In this study, the regions of an utterance that are highly influenced by the emotive state of the speaker are detected. These regions are labeled as emotionally significant regions. Data from the detected emotionally significant regions is used for developing an emotion recognition system. Physiological constraints of human speech production system are explored for detecting the emotionally significant regions of an utterance. Spectral features from the detected emotionally significant regions are used to develop an emotion recognition system. A significant improvement in the performance of the emotion recognition system is observed inthe present approach. An average improvement of 11% is in noted owing to the use of data from emotionally significant regions while developing an emotion recognition system. Gaussian mixture modelling (GMM) technique is employed to develop an emotion recognition system. During the present study, speech samples from Berlin emotion speech database (EMO-DB) are used. Four basic emotions such as anger, happy, neutral and fear are considered for study

Abstract

Query-by-example spoken term detection (QbE-STD) refers to the task of determining the subsequence of a reference which matches with a query, where both the queryand the reference are in audio format. Dynamic time warping(DTW) based techniques are explored to match the two sequences with different lengths in an unsupervised manner. In this paper,a completely unsupervised approach based on Segmental DTW(SDTW), a variant of DTW, is considered for the task of QbE-STD where both reference and query utterances are represented using a sequence of Gaussian posteriorgram vectors. SDTW using two different types of bands i.e., Sakoe-Chiba band and Itakura parallelogram is considered to compare the Gaussian posterior-grams of the query and the reference sequence. The effect o fvarying different local constraints of the DTW algorithm on the performance of SDTW is also analyzed in this paper . Results obtained on MediaEval 2012 dataset indicate that SDTW using a band with variable speaking rate, as in Itakura parallelogram,performs better compared to that of using a band with fixed speaking rate, as in Sakoe-Chiba band, across all variations in local constraints.

Abstract

Multilingual spoken dialogue systems have gained prominence in the recent past necessitating the requirement for a front-end Language Identification (LID) system. Most of the existing LID systems rely on modeling the language discriminative information from low-level acoustic features. Due tothe variabilities of speech (speaker and emotional variabilities, etc.), large-scale LID systems developed using low-levelacoustic features suffer from a degradation in the performance. In this approach, we have attempted to model the higher level language discriminative phonotactic information for developing an LID system. In this paper, the input speech signal is tokenized to phone sequences by using a language independent phone recognizer. The language discriminative phonotactic information in the obtained phone sequences are modeled using statistical and recurrent neural network based language modeling approaches. As this approach, relies on higher level phonotactical information it is more robust to variabilities of speech. Proposed approach is computationally light weight, highly scalable and it can be used in complement with the existing LID systems

Abstract

Most of the existing vowel onset point (VOP) detection methods are developed for clean speech. In this chapter, we propose methods for detection of VOPs in the presence of speech coding and background noise conditions. VOP detection method for coded speech is based on the spectral energy between 500 and 2,500 Hz frequency band of the speech segments present in glottal closure region. In case of noisy speech, the proposed VOP detection method exploits the spectral energy at the formant locations of the speech segments present in glottal closure region. The proposed VOP detection methods are evaluated using objective measures and consonant vowel (CV) unit recognition experiments.

Abstract

This paper proposes a method for automatic detection of breathy voiced vowels in continuous Gujarati speech. As breathy voice is a specific phonetic feature predominantly present in Gujarati among Indian languages, it can be used for identifying Gujarati language. The objective of this paper is to differentiate breathy voiced vowels from modal voiced vowels based on loudness measure. Excitation source characteristics represented by loudness measure are used for differentiating the voice quality. In the proposed method, initially vowel regions in continuous speech are determined by using the knowledge of vowel onset points and epochs. Later, hypothesized vowel segments are classified by using loudness measure. Performance of the proposed method is evaluated on Gujarati speech utterances containing around 47 breathy and 192 modal vowels spoken by 5 male and 5 female speakers. Classification of vowels into breathy or modal voice is achieved with an accuracy of around 94 %.

Abstract

In this paper, the non-uniform duration modification is exploited along with other prosody features for neutral speech to anger speech conversion. The non-uniform duration modification method modifies the durations of vowel and pause segments by different modification factors. Vowel segments are modified by factors based on their identities, and pause segments by uniform factors. Consonant and transition segments are not modified. These modification factors are derived from the analysis of neutral and anger speech. For this purpose, a well known Indian database named as the Indian Institute of Technology Kharagpur Simulated Emotion Speech Corpus (IITKGP-SESC) is chosen for analysis of emotions and synthesis of emotions from neutral speech. The prosodic features used in this study for emotion conversion are pitch contour, intensity contour, and duration contour. Subjective listening test results show that the effectiveness of perception of emotion is better in case of non-uniform duration modification than uniform duration modification