Abstract

With the growing popularity of social networks, cloud services and online applications, people are becoming concerned about the way companies store their data and the ways in which the data can be applied. Privacy with devices and services operated by the voice are of particular interest. To enable studies in privacy, this paper presents a database which quantifies the experience of privacy users have in spoken communication. We focus on the effect of the acoustic environment on that perception of privacy. Speech signals are recorded in scenarios simulating real-life situations, where the acoustic environment has an effect on the experience of privacy. The acoustic data is complemented with measures of the speakers’ experience of privacy, recorded using a questionnaire. The presented corpus enables studies in how acoustic environments affect peoples’ experience of privacy, which in turn, can be used to develop speech operated applications which are respectful of their right to privacy. Index Terms: Experience of privacy, speech interfaces, speech corpus, acoustic environment, right to privacy

Abstract

Majority of the automatic speech recognition systems (ASR) are trained with neutral speech and the performance of these systems are affected due to the presence of emotional content in the speech. The recognition of these emotions in human speech is considered to be the crucial aspect of human-machine interaction. The combined spectral and differenced prosody features are considered for the task of the emotion recognition in the first stage. The task of emotion recognition does not serve the sole purpose of improvement in the performance of an ASR system. Based on the recognized emotions from the input speech, the corresponding adapted emotive ASR model is selected for the evaluation in the second stage. This adapted emotive ASR model is built using the existing neutral and synthetically generated emotive speech using prosody modification method. In this work, the importance of emotion recognition block at the front-end along with the emotive speech adaptation to the ASR system models were studied. The speech samples from IIIT-H Telugu speech corpus were considered for building the large vocabulary ASR systems. The emotional speech samples from IITKGP-SESC Telugu corpus were used for the evaluation. The adapted emotive speech models have yielded better performance over the existing neutral speech models.

Abstract

The easy implementation of replay attacks by a fraudster poses a severe threat to automatic speaker verification (ASV) technology than the other spoofing attacks like speech synthesis and voice conversion. Replay attacks refer to an attack by a fraudster to get illegitimate access to an ASV system by playing back the speech sample collected from genuine target speaker. The significant cues that can differentiate between genuine and replay recordings are channel characteristics. To capture these characteristics, one need to extract features from the spectrum, which should have high spectral and temporal resolutions. Zero time windowing (ZTW) analysis of speech is one such time-frequency analysis technique, which results in high spectral and temporal resolution spectrum at each sampling instant. In this study, new features are proposed by applying cepstral analysis to ZTW spectrum. Experiments are performed on two publicly available replay attack databases namely BTAS 2016 and ASVspoof 2017. The first set of experiments are conducted using Gaussian mixture models to evaluate the potential of proposed features. Performance of the proposed system in terms of half total error rate is 0.75% and in terms of equal error rate is 14.75% on BTAS 2016 and ASVspoof 2017 evaluation sets respectively. A score level fusion is performed by using proposed features with previously proposed single frequency filtering cepstral coefficients. This fused result outperformed the previously reported best results on these two datasets

Abstract

This paper proposes a new speech feature representation that improves the intelligibility assessment of dysarthric speech. The formulation of the feature set is motivated from the human auditory perception and high time-frequency resolution property of single frequency filtering (SFF) technique. The proposed features are named as perceptually enhanced single frequency cepstral coefficients (PESFCC). As a part of SFF technique implementation, speech signal passed through a single pole complex bandpass filter bank to obtain high-resolution time-frequency distribution. Then, the distribution is enhanced by using a set of auditory perceptual operators. Lastly, traditional homomorphic analysis has been carried out on the resulting signal to obtain PE-SFCC feature vector. The performance of proposed features in dysarthric speech detection and its intelligibility assessment has been reported on UASPEECH database. The PE-SFCC features outperformed the state-of-the-art features in dysarthric speech detection and intelligibility assessment.

Abstract

Epochs are the abrupt-closure events in vocal fold vibration during the production of voiced speech. Zero frequency filtering is a simple and effective technique used to estimate the glottal closure instants accurately from the speech signal. However, the zero frequency filter is an unstable system. Hence, it may not be suitable for practical implementation due to the requirements of high precision computation. In this letter, zero-phase zero frequency resonator is proposed as an alternative to zero frequency filter. The proposed approach provides a stable zero-phase response. The experimental results indicate that the performance of the proposed method outperformed the state-of-the-art methods in terms of identification rate 99.17% and provides comparable performance in terms of false alarm rate (0.41%), and identification accuracy (0.28 ms).

Abstract

Self-attention networks are being popularly employed in sequence classification and sequence summarization tasks. State-of-the-art models use sequential models to capture the high-level information, but these models are sensitive to length of utterance and do not equally generalize over variable length utterances. This work explores to study the efficiency of recent advancements in self-attentive networks for improving the performance of the LID system. In self-attentive network, variable length input sequence is converted to fixed dimensional vector which represents the whole sequence. The weighted mean of input sequence is considered as utterance level representation. Along with the mean, a standard deviation is employed to represent the whole input sequence. Experiments are performed using AP17-OLR database. Use of mean with standard deviation has reduced the equal error rate (EER) with an 8% relative improvement. A multi-head attention mechanism is introduced in self-attention networks with an assumption that each head captures the distinct information to discriminate languages. Use of multi-head self-attention has further reduced the EER with a 13% relative improvement. Best performance is achieved with multi-head self-attention network with residual connections. Shifted delta cepstral features (SDC) and stacked SDC features are used for developing LID systems.

Abstract

—India is a multilingual society having more than 1600 languages. Most of these languages are having an overlapping set of phonemes. This makes developing language identification (LID) framework difficult for Indian languages. In this paper, the above challenge is addressed using phonetic features. To model the temporal variations in phonetic features, attention based residual-time delay neural network (RES-TDNN) is proposed. This network effectively captures long-range temporal dependencies through TDNN and attention mechanism. The proposed network has been evaluated on IIITH-ILSC database using phonetic and acoustic features. The database consists of 22 official Indian languages and Indian English. Attention based RES-TDNN outperformed the other state-of-the-art networks such as deep neural network, long short-term memory network and produced an equal error rate of 9.46%. Further, the fusion of shifted delta cepstral and phonetic features have improved the performance.

Abstract

—India is a multilingual society having more than 1600 languages. Most of these languages are having an overlapping set of phonemes. This makes developing language identification (LID) framework difficult for Indian languages. In this paper, the above challenge is addressed using phonetic features. To model the temporal variations in phonetic features, attention based residual-time delay neural network (RES-TDNN) is proposed. This network effectively captures long-range temporal dependencies through TDNN and attention mechanism. The proposed network has been evaluated on IIITH-ILSC database using phonetic and acoustic features. The database consists of 22 official Indian languages and Indian English. Attention based RES-TDNN outperformed the other state-of-the-art networks such as deep neural network, long short-term memory network and produced an equal error rate of 9.46%. Further, the fusion of shifted delta cepstral and phonetic features have improved the performance.

Abstract

In this paper, phonetic features derived from the joint acoustic model (JAM) of a multilingual end to end automatic speech recognition system are proposed for Indian language identification (LID). These features utilize contextual information learned by the JAM through long short-term memoryconnectionist temporal classification (LSTM-CTC) framework. Hence, these features are referred to as CTC features. A multi-head self-attention network is trained using these features, which aggregates the frame-level features by selecting prominent frames through a parametrized attention layer. The proposed features have been tested on IIITH-ILSC database that consists of 22 official Indian languages and Indian English. Experimental results demonstrate that CTC features outperformed i-vector and phonetic temporal neural LID systems and produced an 8.70% equal error rate. The fusion of shifted delta cepstral and CTC feature-based LID systems at the model level and feature level further improved the performance.

Abstract

India being a multilingual society, a multilingual automatic speech recognition system (ASR) is widely appreciated. Despite different orthographies, Indian languages share same phonetic space. To exploit this property, a joint acoustic model has been trained for developing multilingual ASR system using a common phone-set. Three Indian languages namely Telugu, Tamil and, Gujarati are considered for the study. This work studies the amenability of two different acoustic modeling approaches for training a joint acoustic model using common phone-set. Sub-space Gaussian mixture models (SGMM), and recurrent neural networks (RNN) trained with connectionst temporal classification (CTC) objective function are explored for training joint acoustic models. From the experimental results, it can be observed that the joint acoustic models trained with RNN-CTC have performed better than SGMM system even on 120 hours of data (approx 40 hrs per language). The joint acoustic model trained with RNN-CTC has performed better than monolingual models, due to an efficient data sharing across the languages. Conditioning the joint model with language identity had a minimal advantage. Sub-sampling the features by a factor of 2 while training RNN-CTC models has reduced the training times and has performed better. Index Terms: Speech recognition, Joint acoustic model, lowresource, common phone set, Indian languages, RNN-CTC, SGMM