Abstract

Modelling Indian languages is difficult as the number of word forms they present is high. In such cases, prior work has proposed us- ing subwords instead of words as the basic units to model the language. This provides the potential to cover more word forms. Additionally, previous work revealed that all Indian languages have several phonetic similarities. Consequently, multilingual acoustic models have been proposed to counter the data scarcity issue most Indian language datasets have. However, these models use monolin- gual language models (LM). They do not exploit the common basic tokens that certain Indian languages have. This motivated us to im- plement a bilingual subword-based Automatic Speech Recognition (ASR) system for Hindi and Marathi. Further, we try a combination of word-based monolingual LM with bilingual acoustic model to examine the reason for degradation in word-based multilingual ASRs. Our experiments show that multilingual subword-based ASR models outperform their word-based counterparts by upto 9.77% and 26.35% relative Character Error Rate (CER) in the case of Hindi and Marathi respectively.

Abstract

Epoch location is an important parameter for analysis of excitation source information from the speech signals. From several years lot of research have been done to find accurate locations of epochs. Epochs are instants at which vocal tract system are excited signifi- cantly. The prominent location of epochs can be found during the production of speech signals. However, due to the time varying nature of excitation source and the vocal tract system it is difficult to find accurate location of epochs. From various epoch extraction methods, Zero Frequency Filtering (ZFF) is the simplest and most widely used method to find accurate locations of epochs due to its highest identification rate and lowest false alarm rate among other algorithms. ZFF uses Infinite Impulse Response (IIR) filter followed by trend removal blocks however, the filter used in ZFF method is unstable which makes it unsuitable for practical implementation. In the literature many stable implementations of ZFF algorithms have been proposed. Compared to other stable algorithms of ZFF, Zero-phase Zero Frequency Resonator has simple design and gives the highest identification rate among other epoch extraction algo- rithms including ZFF. In this paper, the implementation of ZFF and ZP-ZFR has been proposed on FPGA board using Verilog which is Hardware Description Language (HDL).

Abstract

In this paper, we investigate the use of shifted delta cepstral (SDC) coefficients for detecting disfluencies in two types of speech - stuttered speech and spontaneous lecture-mode speech. SDC features capture temporal variations in the speech signal effectively across several frames. The UCLASS stuttered speech dataset and IIITH-IED dataset are used here to develop frame-level automatic disfluency detection systems for four types of disfluencies and the effect of SDC features on the detection of each disfluency type is examined using MFCC and SFFCC cepstral representations. Overall, it is found that using MFCC+SDC features gives an absolute improvement of 2.98% and 6.02% for stuttered and spontaneous speech disfluency detection respectively over the static MFCC features, while SFFCC+SDC features give an absolute improvement of 4.62% for stutter disfluencies and 7.28% for spontaneous speech disfluencies over the static SFFCC features, showing the importance of considering temporal variations for disfluency detection.

Abstract

Speech systems are sensitive to accent variations. This is especially challenging in the Indian context, with an abundance of languages but a dearth of linguistic studies characterising pronunciation variations. The growing number of L2 English speakers in India reinforces the need to study accents and L1-L2 interactions. We investigate the accents of Indian English (IE) speakers and report in detail our observations, both specific and common to all regions. In particular, we observe the phonemic variations and phonotactics occurring in the speakers' native languages and apply this to their English pronunciations. We demonstrate the influence of 18 Indian languages on IE by comparing the native language pronunciations with IE pronunciations obtained jointly from existing literature studies and phonetically annotated speech of 80 speakers. Consequently, we are able to validate the intuitions of Indian language influences on IE pronunciations by justifying pronunciation rules from the perspective of Indian language phonology. We obtain a comprehensive description in terms of universal and region-specific characteristics of IE, which facilitates accent conversion and adaptation of existing ASR and TTS systems to different Indian accents.

Abstract

Electroencephalography (EEG) provides the temporal resolution required to map the neural activations for studying motor movements and control. This study aims to compare the power amplitude of electrodes covering the central and frontal regions of a 32-channel scalp EEG. The activations from a standard index finger-tapping and a game paradigm are analyzed. Twenty-five right-handed and five left-handed healthy subjects (range = 18–30 years; mean = 24.25 years; SD = 3.96 years) participated in this study. A novel single-frequency filter (SFF) bank was applied to identify the peak amplitude from the power spectral density plots. The results show that the gaming paradigm yields lower or comparable power amplitudes than the standard finger tapping. We observed that the right-hand finger tapping by the left-handed subjects shows lower between-subject dispersion in amplitude. Nonparametric Spearman correlation showed no association between game scores and power amplitude for the right-handed participants. However, for left-handers, both positive and negative associations were observed. This study demonstrates the efficacy of SFF for extracting power amplitudes with a better signalto-noise ratio, which has implications in BCI and motor rehabilitation applications. The findings support the role of game paradigms for motor movement research and in understanding bilateral hemispheric activations in cognitive tasks.

Abstract

In this paper, the intrinsic characteristics of speech modulations are estimated to propose the instant modulation spectral features for efficient emotion recognition. This feature representation is based on single frequency filtering (SFF) technique and higher order nonlinear energy operator. The speech signal is decomposed into frequency sub-bands using SFF, and associated nonlinear energies are estimated with higher order nonlinear energy operator. Then, the feature vector is realized using cepstral analysis. The high-resolution property of SFF technique is exploited to extract the amplitude envelope of the speech signal at a selected frequency with good time-frequency resolution. The fourth order nonlinear energy operator provides noise robustness in estimating the modulation components. The proposed feature set is tested for the emotion recognition task using the i-vector model with the probabilistic linear discriminant scoring scheme, support vector machine and random forest classifiers. The results demonstrate that the performance of this feature representation is better than the widely used spectral and prosody features, achieving detection accuracy of 85.75%, 59.88%, and 65.78% on three emotional databases, EMODB, FAUAIBO, and IEMOCAP, respectively. Further, the proposed features are found to be robust in the presence of additive white Gaussian and vehicular noises.

Abstract

Speech-to-Speech Machine Translation (SSMT) applications and services use a three-step process. Speech recognition is the first step to obtain transcriptions. This is followed by text-to-text language translation and, finally, synthesis into text-speech. As data availability and computing power improved, these individual steps evolved. However, despite significant progress, there is always the error of the first stage in terms of speech recognition, accent, etc. Having traversed the speech recognition stage, the error becomes more prevalent and decreases very often. This chapter presents a complete pipeline for transferring speaker intent in SSMT involving humans in the loop. Initially, the SSMT pipeline has been discussed and analyzed for broadcast speeches and talks on a few sessions of Mann Ki Baat, where the source language is in Hindi, and the target language is in English and Telugu. To perform this task, industry-grade APIs from Google, Microsoft, CDAC, and IITM has been used for benchmarking. Later challenges faced while building the pipeline are discussed, and potential solutions have been introduced. Later this chapter introduces a framework developed to

Abstract

Speech is a natural mode of communication among all beings. India is a densely populated country, and people are diverse throughout the globe. The spoken language is the medium of instruction to interact among the people. The majority of Indian languages are spoken globally. The unavailability of larger volumes of transcribed and annotated speech data is often a hurdle for building reliable speech recognition (ASR) systems for Indian languages. Crowdsourcing strategies are effective in collaboratively collecting speech data resources. This paper describes the experience of large-scale speech data collection for the Telugu language through mobile and web-based applications. With this crowd contributed speech, the performance of the baseline ASR system is shown for clean speech. ASR performance for pink and white noises is also compared for various deep neural network (DNN) based acoustic models. The details regarding the usage of frameworks and their challenges during their implementation are part of this paper. The framework adopted for collecting the speech data is rapid, cost-saving, and offers the advantage of extending it to all the other Indian languages. Index Terms—ASR, Crowd-sourced, TDNN, GMM, SGMM

Abstract

Objective assessment of voice disorders is widely explored as an early diagnosis tool for the classification of voice disorders. Voice disorders affect the pitch, loudness and voice quality, which are perceived at the suprasegmental-level in the speech signal. For the detection and assessment of voice disorders, this study explores the effectiveness of Long Term Average Spectral (LTAS) features using four state-of-the-art filter banks designed with critical-band, constant-Q, gammatone, and singlefrequency filtering approaches. Moreover, the performance of the systems is compared with state-of-the-art statistical-average and openSMILE features. Voice disorder detection experiment was carried out on SVD and HUPA database, while only SVD database is used for assessment task. Assessment task is performed in clinical way, in which four binary classifiers were trained in our study. Voice disorder detection and assessment tasks were carried out using the support vector machine classifier. From the results, it was observed that constant-Q filter bank based LTAS features performed better among all LTAS features with classification accuracy of 78% and 81.4% for voice disorder detection task on SVD and HUPA database, respectively. Further, the combination of LTAS features with OpenSMILE features improved (89.6% and 86.6% for SVD and HUPA database, respectively) the performance.

Abstract

In this paper, a new method to detect disfluencies directly from speech is explored. The method makes use of pre-linguistic automatic syllabification - the process of segmenting input speech signals into perceptually distinct syllable-like regions, to develop syllable-level disfluency detection systems. Statistical prosody features related to fundamental frequency, energy and duration are extracted from each syllable-like region and used to train a DNN classifier for automatic detection of speech disfluencies. Further, complementary information useful for the task of disfluency detection is added to the pipeline with the help of acoustic features. A BiLSTM feature extractor is used to get complex acoustic representation from the baseline MFCC features for each syllable-like region. This acoustic representation is concatenated with the prosody features and used in the proposed system for detecting multiple speech disfluencies. Experiments are conducted for four types of disfluencies in the UCLASS and the IIITH-IED datasets to test the proposed disfluency detection system. Overall, it is found that the proposed system gives a detection accuracy of 88.75% for the disfluencies in the UCLASS dataset, whereas for the IIITH-IED dataset, the accuracy obtained is 91.24%, showing the effectiveness of considering perceptually distinct syllable-like regions as representational units for detecting disfluencies.