Speech recognition with auxiliary information

Automatic speech recognition (ASR) is a very challenging problem due to the wide variety of the data that it must be able to deal with. Being the standard tool for ASR, hidden Markov models (HMMs) have proven to work well for ASR when there are controls over the variety of the data. Being relatively new to ASR, dynamic Bayesian networks (DBNs) are more generic models with algorithms that are more flexible than those of HMMs. Various assumptions can be changed without modifying the underlying algorithm and code, unlike in HMMs; these assumptions relate to the variables to be modeled, the statistical dependencies between these variables, and the observations which are available for certain of the variables. The main objective of this thesis, therefore, is to examine some areas where DBNs can be used to change HMMs' assumptions so as to have models that are more robust to the variety of data that ASR must deal with. HMMs model the standard observed features by jointly modeling them with a hidden discrete state variable and by having certain restraints placed upon the states and features. Some of the areas where DBNs can generalize this modeling framework of HMMs involve the incorporation of even more "auxiliary" variables to help the modeling which HMMs typically can only do with the two variables under certain restraints. The DBN framework is more flexible in how this auxiliary variable is introduced in different ways. First, this auxiliary information aids the modeling due to its correlation with the standard features. As such, in the DBN framework, we can make it directly condition the distribution of the standard features. Second, some types of auxiliary information are not strongly correlated with the hidden state. So, in the DBN framework we may want to consider the auxiliary variable to be conditionally independent of the hidden state variable. Third, as auxiliary information tends to be strongly correlated with its previous values in time, I show DBNs using discretized auxiliary variables that model the evolution of the auxiliary information over time. Finally, as auxiliary information can be missing or noisy in using a trained system, the DBNs can do recognition using just its prior distribution, learned on auxiliary information observations during training. I investigate these different advantages of DBN-based ASR using auxiliary information involving articulator positions, estimated pitch, estimated rate-of-speech, and energy. I also show DBNs to be better at incorporating auxiliary information than hybrid HMM/ANN ASR, using artificial neural networks (ANNs). I show how auxiliary information is best introduced in a time-dependent manner. Finally, DBNs with auxiliary information are better able than standard HMM approaches to handling noisy speech; specifically, DBNs with hidden energy as auxiliary information -- that conditions the distribution of the standard features and which is conditionally independent of the state -- are more robust to noisy speech than HMMs are

Anchored Speech Recognition with Neural Transducers

Neural transducers have achieved human level performance on standard speech recognition benchmarks. However, their performance significantly degrades in the presence of cross-talk, especially when the primary speaker has a low signal-to-noise ratio. Anchored speech recognition refers to a class of methods that use information from an anchor segment (e.g., wake-words) to recognize device-directed speech while ignoring interfering background speech. In this paper, we investigate anchored speech recognition to make neural transducers robust to background speech. We extract context information from the anchor segment with a tiny auxiliary network, and use encoder biasing and joiner gating to guide the transducer towards the target speech. Moreover, to improve the robustness of context embedding extraction, we propose auxiliary training objectives to disentangle lexical content from speaking style. We evaluate our methods on synthetic LibriSpeech-based mixtures comprising several SNR and overlap conditions; they improve relative word error rates by 19.6% over a strong baseline, when averaged over all conditions.Comment: To appear at IEEE ICASSP 202

Modelling auxiliary information (pitch frequency) in hybrid HMM/ANN based ASR systems

Automatic Speech Recognition systems typically use smoothed spectral features as acoustic observations. In recent studies, it has been shown that complementing these standard features with auxiliary information could improve the performance of the system. The previously proposed systems have been studied in the framework of GMMs. In this paper, we study and compare different ways to include auxiliary information in state-of-the-art hybrid HMM/ANN system. In the present paper, we have focused on pitch frequency as the auxiliary information. We have evaluated the proposed system on two different ASR tasks, namely, isolated word recognition and connected word recognition. Our results complement the previous efforts to incorporate auxiliary information in ASR system and also show that pitch frequency can indeed be used in ASR systems to improve the recognition performance

Weakly-Supervised Speech Pre-training: A Case Study on Target Speech Recognition

Self-supervised learning (SSL) based speech pre-training has attracted much attention for its capability of extracting rich representations learned from massive unlabeled data. On the other hand, the use of weakly-supervised data is less explored for speech pre-training. To fill this gap, we propose a weakly-supervised speech pre-training method based on speaker-aware speech data. It adopts a similar training procedure to the widely-used masked speech prediction based SSL framework, while incorporating additional target-speaker enrollment information as an auxiliary input. In this way, the learned representation is steered towards the target speaker even in the presence of highly overlapping interference, allowing potential applications to tasks such as target speech recognition. Our experiments on Libri2Mix and WSJ0-2mix datasets show that the proposed model achieves significantly better ASR performance compared to WavLM, the state-of-the-art SSL model with denoising capability.Comment: Accepted by Interspeech; 5 pages, 1 figure, 3 table

Multiple-Regression Hidden Markov Model

This paper proposes a new class of hidden Markov model (HMM) called multiple-regression HMM (MRHMM) that utilizes auxiliary features such as fundamental frequency (F0) and speaking styles that affect spectral parameters to better model the acoustic features of phonemes. Though such auxiliary features are considered to be the factors that degrade the performance of speech recognizers, the proposed MR-HMM adapts its model parameters, i.e. mean vectors of output probability distributions, depending on these auxiliary information to improve the recognition accuracy. Formulation for parameter reestimation of MRHMM based on the EM algorithm is given in the paper. Experiments of speaker-dependent isolated word recognition demonstrated that MR-HMMs using F0 based auxiliary features reduced the error rates by more than 20% compared with the conventional HMMs

Modeling Auxiliary Information in Bayesian Network Based ASR

Automatic speech recognition bases its models on the acoustic features derived from the speech signal. Some have investigated replacing or supplementing these features with information that can not be precisely measured (articulator positions, pitch, gender, etc.) automatically. Consequently, automatic estimations of the desired information would be generated. This data can degrade performance due to its imprecisions. In this paper, we describe a system that treats pitch as an auxiliary information within the framework of Bayesian networks, resulting in improved performance

Deep Learning Based Speech Enhancement and Its Application to Speech Recognition

Speech enhancement is the task that aims to improve the quality and the intelligibility of a speech signal that is degraded by ambient noise and room reverberation. Speech enhancement algorithms are used extensively in many audio- and communication systems, including mobile handsets, speech recognition, speaker verification systems and hearing aids. Recently, deep learning has achieved great success in many applications, such as computer vision, nature language processing and speech recognition. Speech enhancement methods have been introduced that use deep-learning techniques, as these techniques are capable of learning complex hierarchical functions using large-scale training data. This dissertation investigates the deep learning based speech enhancement and its application to robust Automatic Speech Recognition (ASR). We start our work by exploring generative adversarial network (GAN) based speech enhancement. We explore the techniques to extract information about the noise to aid in the reconstruction of the speech signals. The proposed framework, referred to as ForkGAN, is a novel general adversarial learning-based framework that combines deep-learning with conventional noise reduction techniques. We further extend ForkGAN to M-ForkGAN, which integrates feature mapping and mask learning into a unified framework using ForkGAN. Another variant of ForkGAN, named S-ForkGAN, operates on spectral-domain features, which could directly apply to ASR. Systematic evaluations demonstrate the effectiveness of the proposed approaches. Then, we propose a novel multi-stage learning speech enhancement system. Each stage comprises a self-attention (SA) block followed by stacks of temporal convolutional network (TCN) blocks with doubling dilation factors. Each stage generates a prediction that is refined in a subsequent stage. A fusion block is inserted at the input of later stages to re-inject original information. Moreover, we design several multi-scale architectures with perceptual loss. Experiments show that our proposed architectures can achieve the state of the art performance on several public datasets. Recently, modeling to learn the acoustic noisy-clean speech mapping has been enhanced by including auxiliary information such as visual cues, phonetic and linguistic information, and speaker information. We propose a novel speaker-aware speech enhancement (SASE) method that extracts speaker information from a clean reference using long short-term memory (LSTM) layers, and then uses a convolutional recurrent neural network (CRN) to embed the extracted speaker information. The SASE framework is extended with a self-attention mechanism. It is shown that a few seconds of clean reference speech is sufficient, and that the proposed SASE method performs well for a wide range of scenarios. Even though speech enhancement methods that are based on deep learning have demonstrated state-of-the-art performance when compared with conventional methodologies, current deep learning approaches heavily rely on supervised learning, which requires a large number of noisy- and clean-speech sample pairs for training. This is generally not practical in a realistic environment. One cannot simultaneously obtain both noisy and clean speech samples. Thus, most speech enhancement approaches are trained with simulated speech and clean targets. In addition, it would be hard to collect large-scale dataset for the low-resource languages. We propose a novel noise-to-noise speech enhancement (N2N-SE) method that addresses the parallel noisy-clean training data issue, we leverage signal reconstruction techniques by only using corrupted speech. The proposed N2N-SE framework includes a noise conversion module that is an auto-encoder that learns to mix noise with speech, and a speech enhancement module, that learns to reconstruct corrupted speech signals. In addition to additive noise, speech is also affected by reverberation, which is caused by the attenuated and delayed reflections of sound waves. These distortions, particularly when combined, can severely degrade speech intelligibility for human listeners and impact applications, e.g., automatic speech recognition (ASR) and speaker recognition. Thus, effective speech denoising and dereverberation will benefit both speech processing applications and human listeners. We investigate the deep-learning based approaches for both speech dereverberation and speech denoising using the cascade Conformer architecture. The experimental results show that the proposed cascade Conformer can be effective to suppress the noise and reverberation