Kartik Audhkhasi

Kartik Audhkhasi is a Research Scientist in the Speech Group at Google. His research interests include automatic speech recognition, neural networks, and machine learning. Kartik received his Ph.D. in Electrical Engineering from University of Southern California, USA in 2014. He was a Research Staff Member in the Speech Group at IBM Research AI from 2014 to 2020.

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    Preview abstract Hard and soft distillation are two popular approaches for knowledge distillation from a teacher to student ASR model. Despite soft distillation being better than hard distillation, it has several limitations. First, training convergence depends on the match between the teacher and student alignments. Second, soft distillation suffers quality regressions when using teacher and student models with different architectures. Third, in case of non-causal teacher models, soft distillation requires tuning of the shift in teacher alignments to the right. Finally, soft distillation requires both the teacher and student models to have the same temporal sampling rates. In this work, we propose a novel knowledge distillation method for RNN-T models that tackles limitations of both hard and soft distillation approaches. We call our method Full-sum distillation, which simply distills the sequence posterior probability of the teacher model to the student model. Thus, this method does not depend directly on the noisy labels to distill knowledge as well as it does not depend on time dimension. We also propose a variant of Full-sum distillation to distill the sequence discriminative knowledge of the teacher model to the student model to further improve performance. Using full-sum distillation, we achieve significant improvements when training with strong and weak teacher models on public data as well as on in-house production data. View details
    Preview abstract Federated learning mitigates the need to store user data in a central datastore for machine learning tasks, and is particularly beneficial when working with sensitive user data or tasks. Although successfully used for applications such as improving keyboard query suggestions, it is not studied systematically for modeling affective computing tasks which are often laden with subjective labels and high variability across individuals/raters or even by the same participant. In this paper, we study the federated averaging algorithm FedAvg to model self-reported emotional experience and perception labels on a variety of speech, video and text datasets. We identify two learning paradigms that commonly arise in affective computing tasks: modeling of selfreports (user-as-client), and modeling perceptual judgments such as labeling sentiment of online comments (rater-as-client). In the user-as-client setting, we show that FedAvg generally performs on-par with a non-federated model in classifying self-reports. In the rater-as-client setting, FedAvg consistently performed poorer than its non-federated counterpart. We found that the performance of FedAvg degraded for classes where the interrater agreement was moderate to low. To address this finding, we propose an algorithm FedRater that learns client-specific label distributions in federated settings. Our experimental results show that FedRater not only improves the overall classification performance compared to FedAvg but also provides insights for estimating proxies of inter-rater agreement in distributed settings. View details
    Mixture Model Attention: Flexible Streaming and Non-Streaming Automatic Speech Recognition
    Bhuvana Ramabhadran
    Pedro Jose Moreno Mengibar
    Tongzhou Chen
    Proceedings of Interspeech, 2021 (to appear)
    Preview abstract Streaming automatic speech recognition (ASR) hypothesizes words as soon as the input audio arrives, whereas non-streaming ASR can potentially wait for the completion of the entire utterance to hypothesize words. Streaming and non-streaming ASR systems have typically used different acoustic encoders. Recent work has attempted to unify them by either jointly training a fixed stack of streaming and non-streaming layers or using knowledge distillation during training to ensure consistency between the streaming and non-streaming predictions. We propose mixture model (MiMo) attention as a simpler and theoretically-motivated alternative that replaces only the attention mechanism, requires no change to the training loss, and allows greater flexibility of switching between streaming and non-streaming mode during inference. Our experiments on the public Librispeech data set and a few Indic language data sets show that MiMo attention endows a single ASR model with the ability to operate in both streaming and non-streaming modes without any overhead and without significant loss in accuracy compared to separately-trained streaming and non-streaming models. View details
    Convolutional dropout and wordpiece augmentation for end-to-end speech recognition
    Bhuvana Ramabhadran
    Hainan Xu
    Yinghui Huang
    Yun Zhu
    ICASSP 2021(2021)
    Preview abstract Regularization and data augmentation are crucial to training end-to-end automatic speech recognition systems. Dropout is a popular regularization technique, which operates on each neuron independently by multiplying it with a Bernoulli random variable. We propose a generalization of dropout, called ``convolutional dropout'', where each neuron's activation is replaced with a randomly-weighted linear combination of neuron values in its neighborhood. We believe that this formulation combines the regularizing effect of dropout with the smoothing effects of the convolution operation. In addition to convolutional dropout, this paper also proposes using random wordpiece segmentations as a data augmentation scheme during training, inspired by results in neural machine translation. We adopt both these methods during the training of transformer-transducer speech recognition models, and show consistent improvements over strong baselines across different languages. View details
    Regularizing Word Segmentation by Creating Misspellings
    Bhuvana Ramabhadran
    Hainan Xu
    Jesse Emond
    Yinghui Huang
    Interspeech 2021(2021) (to appear)
    Preview abstract This work focuses on improving subword segmentation algorithms for end-to-end speech recognition models, and makes two major contributions. Firstly, we propose a novel word segmentation algorithm. The algorithm uses the same vocabulary file generated by a regular wordpiece model, is easily extensible and supports a variety of regularization techniques in the segmentation space, and outperforms the regular wordpiece model. Secondly, we propose a number of novel regularization methods that introduces randomness into the tokenization algorithm, which bring further gains in speech recognition performance. A noteworthy discovery from this work is that creating artificial misspelling in words results in the best performance among all the methods, which could inspire future research for strategies in this area. View details
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