In this paper, we introduce a new feature engineering approach for deep learning-based acoustic modeling, which utilizes input feature contributions. For this purpose, we propose an auxiliary deep neural network (DNN) called a feature contribution network (FCN) whose output layer is composed of sigmoid-based contribution gates. In our framework, the FCN tries to learn element-level discriminative contributions of input features and an acoustic model network (AMN) is trained by gated features generated by element-wise multiplication between contribution gate outputs and input features. In addition, we also propose a regularization method for the FCN, which helps the FCN to activate the minimum number of the gates. The proposed methods were evaluated on the TED-LIUM release 1 corpus. We applied the proposed methods to DNN- and long short-term memory-based AMNs. Experimental results results showed that AMNs with the FCNs consistently improved recognition performance compared with AMN-only frameworks.

Patients with locked-in-syndrome (fully paralyzed but aware) struggle in their life and communication. Providing a level of communication offers these patients a chance to resume a meaningful life. Current brain-computer interface (BCI) communication requires users to build words from single letters selected on a screen, which is extremely inefficient. Faster approaches for their speech communication are highly needed. This project investigated the possibility to decode spoken phrases from non-invasive brain activity (MEG) signals. This direct brain-to-text mapping approach may provide a significantly faster communication rate than current BCIs can provide. We used dynamic time warping and Wiener filtering for noise reduction and then Gaussian mixture model and artificial neural network as the decoders. Preliminary results showed the possibility of decoding speech production from non-invasive brain signals. The best phrase classification accuracy was up to 94.54% from single-trial whole-head MEG recordings.

Silent speech recognition (SSR) converts non-audio information such as articulatory movements into text. SSR has the potential to enable persons with laryngectomy to communicate through natural spoken expression. Current SSR systems have largely relied on speaker-dependent recognition models. The high degree of variability in articulatory patterns across different speakers has been a barrier for developing effective speaker-independent SSR approaches. Speaker-independent SSR approaches, however, are critical for reducing the amount of training data required from each speaker. In this paper, we investigate speaker-independent SSR from the movements of flesh points on tongue and lip with articulatory normalization methods that reduce the inter-speaker variation. To minimize the across-speaker physiological differences of the articulators, we propose Procrustes matching-based articulatory normalization by removing locational, rotational, and scaling differences. To further normalize the articulatory data, we apply feature-space maximum likelihood linear regression and i-vector. In this paper, we adopt a bidirectional long short term memory recurrent neural network (BLSTM) as an articulatory model to effectively model the articulatory movements with long-range articulatory history. A silent speech data set with flesh points was collected using an electromagnetic articulograph (EMA) from twelve healthy and two laryngectomized English speakers. Experimental results showed the effectiveness of our speaker-independent SSR approaches on healthy as well as laryngectomy speakers. In addition, BLSTM outperformed standard deep neural network. The best performance was obtained by BLSTM with all the three normalization approaches combined.

A well-trained acoustic model that effectively captures the characteristics of sound events is a critical factor to develop more reliable system for sound event classification. Deep neural network (DNN) which has an ability to extract discriminative representation of features can be a good candidate for acoustic model of sound events. Compared to other data such as speech or image, the amount of sound database is often insufficient for learning the DNN properly, resulting in overfitting problems. In this paper, we propose a cross-acoustic transfer learning framework that can effectively train the DNN even with insufficient sound data by employing rich speech data. Three datasets are used to evaluate our proposed method; one sound dataset is from Real World Computing Partnership (RWCP) DB and two speech datasets are from Resource Management (RM) and Wall Street Journal (WSJ) DBs. A series of experimental results verify that cross-acoustic transfer learning performs significantly better than the baseline DNN which was trained only from sound data, achieving 26.24% relative classification error rate (CER) improvement over the DNN baseline system.

This paper presents a new method for automatically assessing the speech intelligibility of patients with dysarthria, which is a motor speech disorder impeding the physical production of speech. The proposed method consists of two main steps: feature representation and prediction. In the feature representation step, the speech utterance is converted into a phone sequence using an automatic speech recognition technique and is then aligned with a canonical phone sequence from a pronunciation dictionary using a weighted finite state transducer to capture the pronunciation mappings such as match, substitution, and deletion. The histograms of the pronunciation mappings on a pre-defined word set are used for features. Next, in the prediction step, a structured sparse linear model incorporated with phonological knowledge that simultaneously addresses phonologically structured sparse feature selection and intelligibility prediction is proposed. Evaluation of the proposed method on a database of 109 speakers consisting of 94 dysarthric and 15 control speakers yielded a root mean square error of 8.14 compared to subjectively rated scores in the range of 0 to 100. This is a promising performance in which the system can be successfully applied to help speech therapists in diagnosing the degree of speech disorder.

In this paper, the problem of detecting objectionable sounds, such as sexual screaming or moaning, to classify and block objectionable multimedia content is addressed. Objectionable sounds show distinctive characteristics, such as large temporal variations and fast spectral transitions, which are different from general audio signals, such as speech and music. To represent these characteristics, segment-based two-dimensional Mel-frequency cepstral coefficients and histograms of gradient directions are used as a feature set to characterize the time-frequency dynamics within a long-range segment of the target signal. After extracting the features, they are transformed to features with lower dimensions while preserving discriminative information using linear discriminant analysis based on a combination of global and local Fisher criteria. A Gaussian mixture model is adopted to statistically represent objectionable and non-objectionable sounds, and test sounds are classified by using a likelihood ratio test. Evaluation of the proposed feature extraction method on a database of several hundred objectionable and non-objectionable sound clips yielded precision/recall breakeven point of 91.25%, which is a promising performance which shows that the system can be applied to help an image-based approach to block such multimedia content.

This paper addresses the problem of recognizing malicious sounds, such as sexual scream or moan, to detect and block the objectionable multimedia contents. The malicious sounds show the distinct characteristics that have large temporal variations and fast spectral transitions. Therefore, extracting appropriate features to properly represent these characteristics is important in achieving a better performance. In this paper, we employ segment-based two-dimensional Mel-frequency cepstral coefficients and histograms of gradient directions as a feature set to characterize both the temporal variations and spectral transitions within a long-range segment of the target signal. Gaussian mixture model (GMM) is adopted to statistically represent the malicious and non-malicious sounds, and the test sounds are classified by a maximum a posterior probability (MAP) method. Evaluation of the proposed feature extraction method on a database of several hundred malicious and non-malicious sound clips yielded precision of 91.31% and recall of 94.27%. This result suggests that this approach could be used as an alternative to the image-based methods.