A Feature Engineering Approach for Literary and Colloquial Tamil Speech Classification using 1D-CNN

In ideal human computer interaction (HCI), the colloquial form of a language would be preferred by most users, since it is the form used in their day-to-day conversations. However, there is also an undeniable necessity to preserve the formal literary form. By embracing the new and preserving the old, both service to the common man (practicality) and service to the language itself (conservation) can be rendered. Hence, it is ideal for computers to have the ability to accept, process, and converse in both forms of the language, as required. To address this, it is first necessary to identify the form of the input speech, which in the current work is between literary and colloquial Tamil speech. Such a front-end system must consist of a simple, effective, and lightweight classifier that is trained on a few effective features that are capable of capturing the underlying patterns of the speech signal. To accomplish this, a one-dimensional convolutional neural network (1D-CNN) that learns the envelope of features across time, is proposed. The network is trained on a select number of handcrafted features initially, and then on Mel frequency cepstral coefficients (MFCC) for comparison. The handcrafted features were selected to address various aspects of speech such as the spectral and temporal characteristics, prosody, and voice quality. The features are initially analyzed by considering ten parallel utterances and observing the trend of each feature with respect to time. The proposed 1D-CNN, trained using the handcrafted features, offers an F1 score of 0.9803, while that trained on the MFCC offers an F1 score of 0.9895. In light of this, feature ablation and feature combination are explored. When the best ranked handcrafted features, from the feature ablation study, are combined with the MFCC, they offer the best results with an F1 score of 0.9946.

Development of Large Annotated Music Datasets using HMM-based Forced Viterbi Alignment

Datasets are essential for any machine learning task. Automatic Music Transcription (AMT) is one such task, where considerable amount of data is required depending on the way the solution is achieved. Considering the fact that a music dataset, complete with audio and its time-aligned transcriptions would require the effort of people with musical experience, it could be stated that the task becomes even more challenging. Musical experience is required in playing the musical instrument(s), and in annotating and verifying the transcriptions. We propose a method that would help in streamlining this process, making the task of obtaining a dataset from a particular instrument easy and efficient. We use predefined guitar exercises and hidden Markov model(HMM) based forced viterbi alignment to accomplish this. The guitar exercises are designed to be simple. Since the note sequence are already defined, HMM based forced viterbi alignment provides time-aligned transcriptions of these audio files. The onsets of the transcriptions are manually verified and the labels are accurate up to 10ms, averaging at 5ms. The contributions of the proposed work is two fold, i) a well streamlined and efficient method for generating datasets for any instrument, especially monophonic and, ii) an acoustic plectrum guitar dataset containing wave files and transcriptions in the form of label files. This method will aid as a preliminary step towards building concrete datasets for building AMT systems for different instruments.

Literary and Colloquial Dialect Identification for Tamil using Acoustic Features

The evolution and diversity of a language is evident from it's various dialects. If the various dialects are not addressed in technological advancements like automatic speech recognition and speech synthesis, there is a chance that these dialects may disappear. Speech technology plays a role in preserving various dialects of a language from going extinct. In order to build a full fledged automatic speech recognition system that addresses various dialects, an Automatic Dialect Identification (ADI) system acting as the front end is required. This is similar to how language identification systems act as front ends to automatic speech recognition systems that handle multiple languages. The current work proposes a way to identify two popular and broadly classified Tamil dialects, namely literary and colloquial Tamil. Acoustical characteristics rather than phonetics and phonotactics are used, alleviating the requirement of language-dependant linguistic tools. Hence one major advantage of the proposed method is that it does not require an annotated corpus, hence it can be easily adapted to other languages. Gaussian Mixture Models (GMM) using Mel Frequency Cepstral Coefficient (MFCC) features are used to perform the classification task. The experiments yielded an error rate of 12%. Vowel nasalization, as being the reason for this good performance, is discussed. The number of mixture models for the GMM is varied and the performance is analysed.

MaskCycleGAN-based Whisper to Normal Speech Conversion

Whisper to normal speech conversion is an active area of research. Various architectures based on generative adversarial networks have been proposed in the recent past. Especially, recent study shows that MaskCycleGAN, which is a mask guided, and cyclic consistency keeping, generative adversarial network, performs really well for voice conversion from spectrogram representations. In the current work we present a MaskCycleGAN approach for the conversion of whispered speech to normal speech. We find that tuning the mask parameters, and pre-processing the signal with a voice activity detector provides superior performance when compared to the existing approach. The wTIMIT dataset is used for evaluation. Objective metrics such as PESQ and G-Loss are used to evaluate the converted speech, along with subjective evaluation using mean opinion score. The results show that the proposed approach offers considerable benefits.

Quartered Chirp Spectral Envelope for Whispered vs Normal Speech Classification

Whispered speech as an acceptable form of human-computer interaction is gaining traction. Systems that address multiple modes of speech require a robust front-end speech classifier. Performance of whispered vs normal speech classification drops in the presence of additive white Gaussian noise, since normal speech takes on some of the characteristics of whispered speech. In this work, we propose a new feature named the quartered chirp spectral envelope, a combination of the chirp spectrum and the quartered spectral envelope, to classify whispered and normal speech. The chirp spectrum can be fine-tuned to obtain customized features for a given task, and the quartered spectral envelope has been proven to work especially well for the current task. The feature is trained on a one dimensional convolutional neural network, that captures the trends in the spectral envelope. The proposed system performs better than the state of the art, in the presence of white noise.

Literary and Colloquial Tamil Dialect Identification

Culture and language evolve together. The old literary form of Tamil is used commonly for writing and the contemporary colloquial Tamil is used for speaking. Human-computer interaction applications require Colloquial Tamil (CT) to make it more accessible and easy for the everyday user and, it requires Literary Tamil (LT) when information is needed in a formal written format. Continuing the use of LT alongside CT in computer aided language learning applications will both preserve LT, and provide ease of use via CT, at the same time. Hence there is a need for the conversion between LT and CT dialects, which demands as a first step, dialect identification. Dialect Identification (DID) of LT and CT is an unexplored area of research. In the current work, keeping the nuances of both these dialects in mind, five methods are explored which include two implicit methods - Gaussian Mixture Model (GMM) and Convolutional Neural Network (CNN); two explicit methods - Parallel Phone Recognition (PPR) and Parallel Large Vocabulary Continuous Speech Recognition (P-LVCSR); two versions of the proposed explicit Unified Phone Recognition method (UPR-1 and UPR-2). These methods vary based on: the need for annotated data, the size of the unit, the way in which modelling is carried out, and the way in which the final decision is made. Even though the average duration of the test utterances is less - 4.9s for LT and 2.5s for CT - the systems performed well, offering the following identification accuracies: 87.72% (GMM), 93.97% (CNN), 89.24% (PPR), 94.21% (P-LVCSR), 88.57% (UPR-1), 93.53% (UPR-1 with P-LVCSR), 94.55% (UPR-2), and 95.61% (UPR-2 with P-LVCSR).

Quartered Spectral Envelope and 1D-CNN-based Classification of Normally Phonated and Whispered Speech

Whisper, as a form of speech, is not sufficiently addressed by mainstream speech applications. This is due to the fact that systems built for normal speech do not work as expected for whispered speech. A first step to building a speech application that is inclusive of whispered speech, is the successful classification of whispered speech and normal speech. Such a front-end classification system is expected to have high accuracy and low computational overhead, which is the scope of this paper. One of the characteristics of whispered speech is the absence of the fundamental frequency (or pitch), and hence the pitch harmonics as well. The presence of the pitch and pitch harmonics in normal speech, and its absence in whispered speech, is evident in the spectral envelope of the Fourier transform. We observe that this characteristic is predominant in the first quarter of the spectrum, and exploit the same as a feature. We propose the use of one dimensional convolutional neural networks (1D-CNN) to capture these features from the quartered spectral envelope (QSE). The system yields an accuracy of 99.31% when trained and tested on the wTIMIT dataset, and 100% on the CHAINS dataset. The proposed feature is compared with Mel frequency cepstral coefficients (MFCC), a staple in the speech domain. The proposed classification system is also compared with the state-of-the-art system based on log-filterbank energy (LFBE) features trained on long short-term memory (LSTM) network. The proposed system based on 1D-CNN performs better than, or as good as, the state-of-the-art across multiple experiments. It also converges sooner, with lesser computational overhead. Finally, the proposed system is evaluated under the presence of white noise at various signal-to-noise ratios and found to be robust.

Chirp Group Delay based Onset Detection in Instruments with Fast Attack

The onset of a musical note is the earliest time at which a note can be reliably detected. Detection of these musical onsets pose challenges in the presence of ornamentation such as vibrato, bending, and if the attack of the note transient is slower. The legacy systems such as spectral difference or flux and complex domain functions suffer from the addition of false positives due to ornamentation posing as viable onsets. We propose that this can be solved by appropriately improving the resolution of the onset strength signal (OSS) and smoothening it to increase true positives and decrease false positives, respectively. An appropriate peak picking algorithm that works well in unison with the OSS generated is also desired. Since onset detection is a low-level process upon which many other tasks are built, computational complexity must also be reduced. We propose an onset detection alogrithm that is a combination of short-time spectral average-based OSS estimation, chirp group delay-based smoothening, and valley-peak distance-based peak picking. This algorithm performs on par with the state-of-the-art, superflux and convolutional neural networks-based onset detection, with an average F1 score of 0.88, across three datasets. Subsets from the IDMT-SMT-Guitar, Guitarset, and Musicnet datasets that fit the scope of the work, are used for evaluation. It is also found that the proposed algorithm is computationally 300\% more efficient than superflux. The positive effects of smoothening an OSS, in determining the onset locations, is established by refining the OSS produced by legacy algorithms, where consistent improvement in onset detection performance is observed. To provide insights into the performance of the proposed algorithms when different ornamentation styles are present in the recording, three levels of results are computed, by selecting different subsets of the IDMT dataset.

Significance of Chirp MFCC as a Feature in Speech and Audio Applications

A novel feature, based on the chirp z-transform, that offers an improved representation of the underlying true spectrum is proposed. This feature, the chirp MFCC, is derived by computing the Mel frequency cepstral coefficients from the chirp magnitude spectrum, instead of the Fourier transform magnitude spectrum. The theoretical foundations for the proposal, and the experimental validation using product of likelihood Gaussians, to show the improved class separation offered by the proposed chirp MFCC, when compared with vanilla MFCC are discussed. Further, real world evaluation of the feature is performed using three diverse tasks, namely, speech-music classification, speaker identification, and speech commands recognition. It is shown in all three tasks that the proposed chirp MFCC offers considerable improvements.