A Subband-Based SVM Front-End for Robust ASR

This work proposes a novel support vector machine (SVM) based robust automatic speech recognition (ASR) front-end that operates on an ensemble of the subband components of high-dimensional acoustic waveforms. The key issues of selecting the appropriate SVM kernels for classification in frequency subbands and the combination of individual subband classifiers using ensemble methods are addressed. The proposed front-end is compared with state-of-the-art ASR front-ends in terms of robustness to additive noise and linear filtering. Experiments performed on the TIMIT phoneme classification task demonstrate the benefits of the proposed subband based SVM front-end: it outperforms the standard cepstral front-end in the presence of noise and linear filtering for signal-to-noise ratio (SNR) below 12-dB. A combination of the proposed front-end with a conventional front-end such as MFCC yields further improvements over the individual front ends across the full range of noise levels

A Comparison of Decision Tree Classifiers for Automatic Diagnosis of Speech Recognition Errors

Present speech recognition systems are becoming more complex due to technology advances, optimizations and special requirements such as small computation and memory footprints. Proper handling of system failures can be seen as a kind of fault diagnosis. Motivated by the success of decision tree diagnosis in other scientific fields and by their successful application in speech recognition in the last decade, we contribute to the topic mainly in terms of comparison of different types of decision trees. Five styles are examined: CART (testing three different splitting criteria), C4.5, and then Minimum Message Length (MML), strict MML and Bayesian styles decision trees. We apply these techniques to data of computer speech recognition fed by intrinsically variable speech. We conclude that for this task, CART technique outperforms C4.5 in terms of better classification for ASR failures

Environmentally robust ASR front-end for deep neural network acoustic models

This paper examines the individual and combined impacts of various front-end approaches on the performance of deep neural network (DNN) based speech recognition systems in distant talking situations, where acoustic environmental distortion degrades the recognition performance. Training of a DNN-based acoustic model consists of generation of state alignments followed by learning the network parameters. This paper first shows that the network parameters are more sensitive to the speech quality than the alignments and thus this stage requires improvement. Then, various front-end robustness approaches to addressing this problem are categorised based on functionality. The degree to which each class of approaches impacts the performance of DNN-based acoustic models is examined experimentally. Based on the results, a front-end processing pipeline is proposed for efficiently combining different classes of approaches. Using this front-end, the combined effects of different classes of approaches are further evaluated in a single distant microphone-based meeting transcription task with both speaker independent (SI) and speaker adaptive training (SAT) set-ups. By combining multiple speech enhancement results, multiple types of features, and feature transformation, the front-end shows relative performance gains of 7.24% and 9.83% in the SI and SAT scenarios, respectively, over competitive DNN-based systems using log mel-filter bank features.This is the final version of the article. It first appeared from Elsevier via http://dx.doi.org/10.1016/j.csl.2014.11.00

Developing a Noise-Robust Beat Learning Algorithm for Music-Information Retrieval

The field of Music-Information Retrieval (Music-IR) involves the development of algorithms that can analyze musical audio and extract various high-level musical features. Many such algorithms have been developed, and systems now exist that can reliably identify features such as beat locations, tempo, and rhythm from musical sources. These features in turn are used to assist in a variety of music-related tasks ranging from automatically creating playlists that match specified criteria to synchronizing various elements, such as computer graphics, with a performance. These Music-IR systems thus help humans to enjoy and interact with music. While current systems for identifying beats in music are have found widespread utility, most of them have been developed on music that is relatively free of acoustic noise. Much of the music that humans listen to, though, is performed in noisy environments. People often enjoy music in crowded clubs and noisy rooms, but this music is much more challenging for Music-IR systems to analyze, and current beat trackers generally perform poorly on musical audio heard in such conditions. If our algorithms could accurately process this music, though, it would enable this music too to be used in applications such as automatic song selection, which are currently limited to music taken directly from professionally-produced digital files that have little acoustic noise. Noise-robust beat learning algorithms would also allow for additional types of performance augmentation which create noise and thus cannot be used with current algorithms. Such a system, for instance, could aid robots in performing synchronously with music, whereas current systems are generally unable to accurately process audio heard in conjunction with noisy robot motors. This work aims to present a new approach for learning beats and identifying both their temporal locations and their spectral characteristics for music recorded in the presence of noise. First, datasets of musical audio recorded in environments with multiple types of noise were collected and annotated. Noise sources used for these datasets included HVAC sounds from a room, chatter from a crowded bar, and fans and motor noises from a moving robot. Second, an algorithm for learning and locating musical beats was developed which incorporates signal processing and machine learning techniques such as Harmonic-Percussive Source Separation and Probabilistic Latent Component Analysis. A representation of the musical signal called the stacked spectrogram was also utilized in order to better represent the time-varying nature of the beats. Unlike many current systems, which assume that the beat locations will be correlated with some hand-crafted features, this system learns the beats directly from the acoustic signal. Finally, the algorithm was tested against several state-of-the-art beat trackers on the audio datasets. The resultant system was found to significantly outperform the state-of-the-art when evaluated on audio played in realistically noisy conditions.Ph.D., Electrical Engineering -- Drexel University, 201