Zhong-Qiu Wang

Combining spectral feature mapping and multi-channel model-based source separation for noise-robust automatic speech recognition

Automatic Speech Recognition systems suffer from severe performance degradation in the presence of myriad complicating factors such as noise, reverberation, multiple speech sources, multiple recording devices, etc. Previous challenges have sparked much innovation when it comes to designing systems capable of handling these complications. In this spirit, the CHiME-3 challenge presents system builders with the task of recognizing speech in a real-world noisy setting wherein speakers talk to an array of 6 microphones in a tablet. In order to address these issues, we explore the effectiveness of first applying a model-based source separation mask to the output of a beamformer that combines the source signals recorded by each microphone, followed by a DNN-based front end spectral mapper that predicts clean filterbank features. The source separation algorithm MESSL (Model-based EM Source Separation and Localization) has been extended from two channels to multiple channels in order to meet the demands of the challenge. We report on interactions between the two systems, cross-cut by the use of a robust beamforming algorithm called BeamformIt. Evaluations of different system settings reveal that combining MESSL and the spectral mapper together on the baseline beamformer algorithm boosts the performance substantially.

A joint training framework for robust automatic speech recognition

Robustness against noise and reverberation is critical for ASR systems deployed in real-world environments. In robust ASR, corrupted speech is normally enhanced using speech separation or enhancement algorithms before recognition. This paper presents a novel joint training framework for speech separation and recognition. The key idea is to concatenate a deep neural network (DNN) based speech separation frontend and a DNN-based acoustic model to build a larger neural network, and jointly adjust the weights in each module. This way, the separation frontend is able to provide enhanced speech desired by the acoustic model and the acoustic model can guide the separation frontend to produce more discriminative enhancement. In addition, we apply sequence training to the jointly trained DNN so that the linguistic information contained in the acoustic and language models can be back-propagated to influence the separation frontend at the training stage. To further improve the robustness, we add more noise- and reverberation-robust features for acoustic modeling. At the test stage, utterance-level unsupervised adaptation is performed to adapt the jointly trained network by learning a linear transformation of the input of the separation frontend. The resulting sequence-discriminative jointly-trained multistream system with run-time adaptation achieves 10.63% average word error rate (WER) on the test set of the reverberant and noisy CHiME-2 dataset (task-2), which represents the best performance on this dataset and a 22.75% error reduction over the best existing method.

Phoneme-specific speech separation

Speech separation or enhancement algorithms seldom exploit information about phoneme identities. In this study, we propose a novel phoneme-specific speech separation method. Rather than training a single global model to enhance all the frames, we train a separate model for each phoneme to process its corresponding frames. A robust ASR system is employed to identify the phoneme identity of each frame. This way, the information from ASR systems and language models can directly influence speech separation by selecting a phoneme-specific model to use at the test stage. In addition, phoneme-specific models have fewer variations to model and do not exhibit the data imbalance problem. The improved enhancement results can in turn help recognition. Experiments on the corpus of the second CHiME speech separation and recognition challenge (task-2) demonstrate the effectiveness of this method in terms of objective measures of speech intelligibility and quality, as well as recognition performance.

Learning utterance-level representations for speech emotion and age/gender recognition using deep neural networks

Accurately recognizing speaker emotion and age/gender from speech can provide better user experience for many spoken dialogue systems. In this study, we propose to use deep neural networks (DNNs) to encode each utterance into a fixed-length vector by pooling the activations of the last hidden layer over time. The feature encoding process is designed to be jointly trained with the utterance-level classifier for better classification. A kernel extreme learning machine (ELM) is further trained on the encoded vectors for better utterance-level classification. Experiments on a Mandarin dataset demonstrate the effectiveness of our proposed methods on speech emotion and age/gender recognition tasks.

Recurrent deep stacking networks for supervised speech separation

Supervised speech separation algorithms seldom utilize output patterns. This study proposes a novel recurrent deep stacking approach for time-frequency masking based speech separation, where the output context is explicitly employed to improve the accuracy of mask estimation. The key idea is to incorporate the estimated masks of several previous frames as additional inputs to better estimate the mask of the current frame. Rather than formulating it as a recurrent neural network (RNN), which is potentially much harder to train, we propose to train a deep neural network (DNN) with implicit deep stacking. The estimated masks of the previous frames are updated only at the end of each DNN training epoch, and then the updated estimated masks provide additional inputs to train the DNN in the next epoch. At the test stage, the DNN makes predictions sequentially in a recurrent fashion. In addition, we propose to use the L1 loss for training. Experiments on the CHiME-2 (task-2) dataset demonstrate the effectiveness of our proposed approach.

A two-stage algorithm for noisy and reverberant speech enhancement

In daily listening environments, speech is commonly corrupted by room reverberation and background noise. These distortions are detrimental to speech intelligibility and quality, and also severely degrade the performance of automatic speech and speaker recognition systems. In this paper, we propose a two-stage algorithm to deal with the confounding effects of noise and reverberation separately, where denoising and dereverberation are conducted sequentially using deep neural networks. In addition, we design a new objective function that incorporates clean phase information during training. As the objective function emphasizes more important time-frequency (T-F) units, better estimated magnitude is obtained during testing. By jointly training the two-stage model to optimize the proposed objective function, our algorithm improves objective metrics of speech intelligibility and quality significantly, and substantially outperforms one-stage enhancement baselines.

A speech enhancement algorithm by iterating single- and multi-microphone processing and its application to robust ASR

We propose a speech enhancement algorithm based on single- and multi-microphone processing techniques. The core of the algorithm estimates a time-frequency mask which represents the target speech and use masking-based beamforming to enhance corrupted speech. Specifically, in single-microphone processing, the received signals of a microphone array are treated as individual signals and we estimate a mask for the signal of each microphone using a deep neural network (DNN). With these masks, in multi-microphone processing, we calculate a spatial covariance matrix of noise and steering vector for beamforming. In addition, we propose a masking-based post-filter to further suppress the noise in the output of beamforming. Then, the enhanced speech is sent back to DNN for mask re-estimation. When these steps are iterated for a few times, we obtain the final enhanced speech. The proposed algorithm is evaluated as a frontend for automatic speech recognition (ASR) and achieves a 5.05% average word error rate (WER) on the real environment test set of CHiME-3, outperforming the current best algorithm by 13.34%.

Unsupervised speaker adaptation of batch normalized acoustic models for robust ASR

Batch normalization is a standard technique for training deep neural networks. In batch normalization, the input of each hidden layer is first mean-variance normalized and then linearly transformed before applying non-linear activation functions. We propose a novel unsupervised speaker adaptation technique for batch normalized acoustic models. The key idea is to adjust the linear transformations previously learned by batch normalization for all the hidden layers according to the first-pass decoding results of the speaker-independent model. With the adjusted linear transformations for each test speaker, the test distribution of the input of each hidden layer better matches the training distribution. Experiments on the CHiME-3 dataset demonstrate the effectiveness of the proposed layer-wise adaptation approach. Our overall system obtains 4.24% WER on the real subset of the test data, which represents the best reported result on this dataset to date and a relative 27.3% error reduction over the previous best result.

Speech emotion recognition based on Gaussian Mixture Models and Deep Neural Networks

Recognition of speaker emotion during interaction in spoken dialog systems can enhance the user experience, and provide system operators with information valuable to ongoing assessment of interaction system performance and utility. Interaction utterances are very short, and we assume the speakers emotion is constant throughout a given utterance. This paper investigates combinations of a GMM-based low-level feature extractor with a neural network serving as a high level feature extractor. The advantage of this system architecture is that it combines the fast developing neural network-based solutions with the classic statistical approaches applied to emotion recognition. Experiments on a Mandarin data set compare different solutions under the same or close conditions.

Robust speech recognition from ratio masks

Robustness against noise is crucial for automatic speech recognition systems in real-world environments. In this paper, we propose a novel approach that performs robust ASR by directly recognizing ratio masks. In the proposed approach, a deep neural network (DNN) is first trained to estimate the ideal ratio mask (IRM) from a noisy utterance and then a convolutional neural network (CNN) is employed to recognize estimated IRMs. The proposed approach has been evaluated on the TIDigits corpus, and the results demonstrate that direct recognition of ratio masks outperforms direct recognition of binary masks and traditional MMSE-HMM based method for robust ASR.