Calendar

Oct
17
Thu
Thesis Proposal: Yansong Zhu @ Olin Hall 305
Oct 17 @ 3:00 pm – 4:00 pm

Title: Advanced Image Reconstruction and Analysis for Fluorescence Molecular Tomography (FMT) and Positron Emission Tomography (PET)

Abstract: Molecular imaging provides efficient ways to monitor different biological processes noninvasively, and high-quality imaging is necessary in order to fully explore the value of molecular imaging. To this end, advanced image generation algorithms are able to significantly improve image quality and quantitative performance. In this research proposal, we focus on two imaging modalities, fluorescence molecular tomography (FMT) and positron emission tomography (PET), that fall in the category of molecular imaging. Specifically, we studied the following two problems: i) reconstruction problem in FMT and ii) partial volume correction in brain PET imaging.

Reconstruction in FMT: FMT is an optical imaging modality that uses diffuse light for imaging. Reconstruction problem for FMT is highly ill-posed due to photon scattering in biological tissue, and thus, regularization techniques tend to be used to alleviate the ill-posed nature of the problem. Conventional reconstruction algorithms cause oversmoothing which reduces resolution of the reconstructed images. Moreover, a Gaussian model is commonly chosen as the noise model although most FMT systems based on charged-couple device (CCD) or photon multiplier tube (PMT) are contaminated by Poisson noise. In our work, we propose a reconstruction algorithm for FMT using sparsity-initialized maximum-likelihood expectation maximization (MLEM). The algorithm preserves edges by exploiting sparsity, as well as taking Poisson noise into consideration. Through simulation experiments, we compare the proposed method with pure sparse reconstruction method and MLEM with uniform initialization. We show the proposed method holds several advantages compared to the other two methods.

Partial volume correction of brain PET imaging: The so-called partial volume effect (PVE) is caused by the limited resolution of PET systems, reducing quantitative accuracy of PET imaging. Based on the stage of implementation, partial volume correction (PVC) algorithms could be categorized into reconstruction-based and post-reconstruction methods.Post reconstruction PVC methods can be directly implemented on reconstructed PET images and do not require access to raw data or reconstruction algorithms of PET scanners. Many of these methods use anatomical information from MRI to further improve their performance. However, conventional MR guided post-reconstruction PVC methods require segmentation of MR images and assume uniform activity distribution within each segmented region. In this proposal, we develop post-reconstruction PVC method based on deconvolution via parallel level set regularization. The method is implemented with non-smooth optimization based on the split Bregman method. The proposed method incorporates MRI information without requiring segmentation or making any assumption on activity distribution. Simulation experiments are conducted to compare the proposed method with several other segmentationfree method, as well as conventional segmentation-based PVC method. The results show the proposed method outperforms other segmentation-free method and shows stronger resistance to MR information mismatch compared to conventional segmentation-based PVC method.

Oct
31
Thu
Thesis Proposal: Jordi Abante Llenas @ Olin Hall 305
Oct 31 @ 3:00 pm – 4:00 pm
Thesis Proposal: Jordi Abante Llenas @ Olin Hall 305

Title: Statistical Modeling and analysis of allele-specific DNA methylation at the haplotype level

Abstract: Epigenetics is the branch of biology concerned with the study of phenotypical changes due to alterations of DNA, maintained during cell division, excluding modifications of the sequence itself. Epigenetic information includes DNA methylation, histone modifications, and higher order chromatin structure among others. DNA methylation is a stable epigenetic mechanism that chemically marks the DNA by adding methyl groups at individual cytosines immediately adjacent to guanines (CpG sites). Methylation marks are used to identify cell-type specific aspects of gene regulation, since marks located within a gene promoter or enhancer typically act to repress gene transcription, whereas promoter or enhancer demethylation is associated with gene activation. Notably, patterns of methylation marks are highly polymorphic and stochastic, containing information about a broad range of normal and aberrant biological processes, such as development and differentiation, aging, and carcinogenesis.

The epigenetic information content of two homologous chromosomal regions need not be the same. For example, it is well established that the ability of a cell to methylate the promoter region of a specific copy of a gene (an allele), is crucial for proper development. In fact, many known phenotypical traits stem from allele-specific epigenetic marks. Moreover, some allele-specific epigenetic differences have been found to be associated with local genetic differences between copies of a chromosome. Thus, developing a framework for studying such epigenetic differences in diploid organisms is our main goal. More specifically, our objective is to develop a statistical method that can be used to detect regions in the genome, with genetic differences between homologous chromosomes, in which there are biologically relevant differences in DNA methylation between alleles.

State of the art methods for allele-specific methylation modeling and analysis have critical shortcomings rendering them unsuitable for this type of analysis. We present a statistical physics inspired model for allele-specific methylation analysis that contains a sensible number of parameters, considering the limited sample size in whole genome bisulfite sequencing data, which is rich enough to capture the complexity in the data. We demonstrate the appropriateness of this model for allele-specific methylation analysis using simulation data as well as real data. Using our model, we compute mean methylation level differences between alleles, as well as information-theoretic quantities, such as the entropy of the methylation state in each allele and the mutual information between the methylation state and the allele of origin, and assess the statistical significance of each quantity by learning the null distribution from the data. This complementary set of statistics allows for an unparalleled level of insight in subsequent biological analysis. As a result, the developed framework provides an unprecedented descriptive power to characterize (i) the circumstances under which allele-specific methylation events arise, and (ii) the cis-effect, or lack of thereof, that genetic mutations have on DNA methylation.

Nov
21
Thu
Thesis Proposal: Ruizhi Li @ Olin Hall 305
Nov 21 @ 3:00 pm – 4:00 pm
Thesis Proposal: Ruizhi Li @ Olin Hall 305

Title: A Practical and Efficient Multi-Stream Framework for End-to-End Speech Recognition

Abstract: The multi-stream paradigm in Automatic Speech Recognition (ASR) considers scenarios where parallel streams carry diverse or complementary task-related knowledge. In these cases, an appropriate strategy to fuse streams or select the most informative source is necessary. In recent years, with the increasing use of Deep Neural Networks (DNNs) in ASR, End-to-End (E2E) approaches, which directly transcribe human speech into text, have received greater attention. In this proposal, a multi-stream framework is present based on joint CTC/Attention E2E model, where parallel streams are represented by separate encoders aiming to capture diverse information. On top of the regular attention networks, a secondary stream-fusion network is introduced to steer the decoder toward the most informative encoders.

Two representative framework have been proposed, which are MultiEncoder Multi-Resolution (MEM-Res) and Multi-Encoder Multi-Array (MEM-Array), respectively. Moreover, with an increasing number of streams (encoders) requiring substantial memory and massive amounts of parallel data, a practical two-stage training scheme is further proposed in this work. Experiments are conducted on various corpora including Wall Street Journal (WSJ), CHiME-4, DIRHA and AMI. Compared with the best single-stream performance, the proposed framework has achieved substantial improvement, which also outperforms various conventional fusion strategies.

The future plan aims to improve robustness of the proposed multistream framework. Measuring performance of an ASR system without ground-truth could be beneficial in multi-stream scenarios to emphasize on more informative streams than corrupted ones. In this proposal, four different Performance Monitoring (PM) techniques are investigated. The preliminary results suggest that PM measures on attention distributions and decoder posteriors are well-correlated with true performances. Integration of PM measures and more sophisticated fusion mechanism in multi-stream framework will be the focus for future exploration.

Jan
30
Thu
Thesis Proposal: Pramuditha Perera @ Hackerman Hall B-17
Jan 30 @ 3:00 pm – 4:00 pm
Thesis Proposal: Pramuditha Perera @ Hackerman Hall B-17

Title: Deep Learning-based Novelty Detection

Abstract: In recent years, intelligent systems powered by artificial intelligence and computer vision that perform visual recognition have gained much attention. These systems observe instances and labels of known object classes during training and learn association patterns that can be
used during inference. A practical visual recognition system should first determine whether an observed instance is from a known class. If it is from a known class, then the identity of the instance is queried through classification. The former process is commonly known as novelty detection (or novel class detection) in the literature. Given a set of image instances from known classes, the goal of novelty detection is to determine whether an observed image during inference belongs to one of the known classes.

We consider one-class novelty detection, where all training data are assumed to belong to a single class without any finer-annotations available. We identify limitations of conventional approaches in one-class novelty detection and present a Generative Adversarial Network(GAN) based solution. Our solution is based on learning latent representations of in-class examples using a denoising auto-encoder network. The key contribution of our work is our proposal to explicitly constrain the latent space to exclusively represent the given class. In order to accomplish this goal, firstly, we force the latent space to have bounded support by introducing a tanh activation in the encoder’s output layer. Secondly, using a discriminator in the latent space that is trained adversarially, we ensure that encoded representations of in-class examples resemble uniform random samples drawn from the same bounded space. Thirdly, using a second adversarial discriminator in the input space, we ensure all randomly drawn latent samples generate examples that look real.

Finally, we introduce a gradient-descent based sampling technique that explores points in the latent space that generate potential out-of-class examples, which are fed back to the network to further train it to generate in-class examples from those points. The effectiveness of the proposed method is measured across four publicly available datasets using two one-class novelty detection protocols where we achieve state-of-the-art results.

Feb
27
Thu
Thesis Proposal: Raghavendra Pappagari @ Hackerman Hall B-17
Feb 27 @ 3:00 pm
Thesis Proposal: Raghavendra Pappagari @ Hackerman Hall B-17

Title: Towards a better understanding of spoken conversations: Assessment of sentiment and emotion

Abstract: In this talk, we present our work on understanding the emotional aspects of spoken conversations. Emotions play a vital role in our daily life as they help us convey information impossible to express verbally to other parties.

While humans can easily perceive emotions, these are notoriously difficult to define and recognize by machines. However, automatically detecting the emotion of a spoken conversation can be useful for a diverse range of applications such as human-machine interaction and conversation analysis. In this work, we considered emotion recognition in two particular scenarios. The first scenario is predicting customer sentiment/satisfaction (CSAT) in a call center conversation, and the second consists of emotion prediction in short utterances.

CSAT is defined as the overall sentiment (positive vs. negative) of the customer about his/her interaction with the agent. In this work, we perform a comprehensive search for adequate acoustic and lexical representations.

For acoustic representation, we propose to use the x-vector model, which is known for its state-of-the-art performance in the speaker recognition task. The motivation behind using x-vectors for CSAT is we observed that emotion information encoded in x-vectors affected speaker recognition performance. For lexical, we introduce a novel method, CSAT Tracker, which computes the overall prediction based on individual segment outcomes. Both methods rely on transfer learning to obtain the best performance. We classified using convolutional neural networks combining the acoustic and lexical features. We evaluated our systems on US English telephone speech from call center data. We found that lexical models perform better than acoustic models and fusion of them provided significant gains. The analysis of errors uncovers that the calls where customers accomplished their goal but were still dissatisfied are the most difficult to predict correctly. Also, we found that the customer’s speech is more emotional compared to the agent’s speech.

For the second scenario of predicting emotion, we present a novel approach based on x-vectors. We show that adapting the x-vector model for emotion recognition provides the best-published results on three public datasets.

Mar
5
Thu
Thesis Proposal: Matthew Maciejewski @ Hackerman Hall B-17
Mar 5 @ 3:00 pm
Thesis Proposal: Matthew Maciejewski @ Hackerman Hall B-17

Title: Single-Channel Speech Separation in Noisy and Reverberant Conditions

Abstract: An inevitable property of multi-party conversations is that more than one speaker will end up speaking simultaneously for portions of time. Many speech technologies, such as automatic speech recognition and speaker identification, are not designed to function on overlapping speech and suffer severe performance degradation under such conditions. Speech separation techniques aim to solve this problem by producing a separate waveform for each speaker in an audio recording with multiple talkers speaking simultaneously. The advent of deep neural networks has resulted in strong performance gains on the speech separation task. However, training and evaluation has been nearly ubiquitously restricted to a single dataset of clean, near-field read speech, not representative of many multi-person conversational settings which are frequently recorded on room microphones, introducing noise and reverberation. Due to the degradation of other speech technologies in these sorts of conditions, speech separation systems are expected to suffer a decrease in performance as well.

The primary goal of this proposal is to develop novel techniques to improve speech separation in noisy and reverberant recording conditions. One core component of this work is the creation of additional synthetic overlap corpora spanning a range of more realistic and challenging conditions. The lack of appropriate data necessitates a first step of creating appropriate conditions with which to benchmark the performance of state-of-the-art methods in these more challenging conditions. Another proposed line of investigation is the integration of speech separation techniques with speech enhancement, the task of enhancing a speech signal through the removal of noise or reverberation. This is a natural combination due to similarities in problem formulation and general approach. Finally, we propose an investigation into the effectiveness of speech separation as a pre-processing step to speech technologies, such as automatic speech recognition, that struggle with overlapping speech, as well as tighter integration of speech separation with these “downstream” systems.

Apr
2
Thu
Thesis Proposal: John Franklin
Apr 2 @ 3:00 pm
Thesis Proposal: John Franklin

This presentation will be happening remotely over Zoom. Click this link as early as 15 minutes before the scheduled start time of the presentation to watch in a Zoom meeting.

Meeting ID: 618-589-385
Meeting Password: 261713

Title: Compressive Sensing for Wireless Systems with Massive Antenna Arrays

Abstract: Over the past two decades the world has enjoyed exponential growth in wireless connectivity that has fundamentally changed the way people communicate and has opened the door to limitless new applications. With the advent of 5G, users will now begin to enjoy enhanced mobile broadband links supporting peak rates of over 10 gigabit per second. The 5G capability will also support massive machine type communications and less than one millisecond latency communications to support ultra-reliable low communication. Continuing to achieve greater increases in system capacity requires the continual advancement of new technology to make efficient use of finite spectrum resources.

Researchers have studied Multiple-Input-Multiple-Output (MIMO) communications over the last several decades as a way to increase system capacity. The MIMO channel is composed of multiple transmit (input) antennas and multiple (output) receive antennas. The channel is represented as the impulse response between each transmit and receive antenna pair. In the simplest of channels, the pairwise impulse response reduces to a single coefficient. Many theoretical MIMO results rely on Rayleigh channels featuring independently distributed complex Gaussian variables as channel coefficients.

The concept of Massive MIMO emerged a decade ago and is a leading technology in 5G wireless. Massive MIMO features base stations that have massive antenna arrays that simultaneously service many users. The Massive MIMO array has many more antennas than users. Unlike traditional phased array antennas, Massive MIMO arrays have all (or a large portion of) their antennas connected to receive chains for baseband processing. Successfully decoding each user’s data stream requires estimates of the propagation channel. Channel estimation is usually aided through the use of pilot signals that are known to both the user terminal and the base station. Simultaneously estimating the channel matrix between each user and each antenna in a massive MIMO array creates challenges for pilot sequence design. More channel resources reserved for pilot sequences for channel estimation result in fewer resources for user data.

Several efforts have shown that the mm wave massive MIMO channel exhibits several sparse features. The number of distinct and resolvable paths between a user and a massive MIMO array is generally much less than the number of base station antennas. Early theoretical MIMO work relied on Rayleigh channels as they are useful for closed form solutions. In reality, the Massive MIMO mm wave channel is low rank as it can be modeled by a smaller number of resolvable multipath components. This opens opportunities for new channel estimation techniques using compressive sensing and sparse recovery.

Although Massive MIMO will be featured in future 5G services, there is still much untapped potential. Through developing better channel estimation schemes, additional system throughput can be achieved. This work will consider:

  • Generation of sparse mm Wave channels for analysis
  • Multi-user pilot design approaches for measuring the massive MIMO channel
  • Channel estimates formed through sparse recovery methods
Apr
16
Thu
Thesis Proposal: Golnoosh Kamali
Apr 16 @ 3:00 pm
Thesis Proposal: Golnoosh Kamali

This event will occur remotely in a Zoom meeting at this link. Please do not join the meeting until at least 15 minutes before the presentation is scheduled to start. 

Title: Using Systems Modeling to Localize the Seizure Onset Zone in Epilepsy Patients from Single Pulse Electrical Stimulation Recordings

Abstract: Surgical resection of the seizure onset zone (SOZ) could potentially lead to seizure-freedom in medically refractory epilepsy patients. However, localizing the SOZ can be a time consuming and tedious process involving visual inspection of intracranial electroencephalographic (iEEG) recordings captured during passive patient monitoring. Single pulse electrical stimulation (SPES) is currently performed on patients undergoing invasive EEG monitoring for the main purposes of mapping functional brain networks such as language and motor networks. We hypothesize that evoked responses from SPES can also be used to localize the SOZ as they may express the natural frequencies and connectivity of the iEEG network. To test our hypothesis, we construct patient specific single-input multi-output transfer function models from the evoked responses recorded from eight epilepsy patients that underwent SPES evaluation and iEEG monitoring. Our preliminary results suggest that the stimulation electrodes that produced the highest system gain, as measured by the 𝓗∞ norm, correspond to those electrodes clinically defined in the SOZ in successfully treated patients.

Apr
23
Thu
Thesis Proposal: Aswin Shanmugam Subramanian
Apr 23 @ 3:00 pm
Thesis Proposal: Aswin Shanmugam Subramanian

This presentation will be done remotely. Follow this link for access to the Zoom meeting where it will be taking place. It is advised that you do not log in to the meeting until at least 15 minutes before the presentation’s start time.

Title: A Synergistic Combination of Signal Processing and Deep Learning for Robust Speech Processing

Abstract: When speech is captured with a distant microphone it includes distortions caused by noise, reverberation and overlapping speakers. Far-field speech processing systems need to be robust to those distortions to function in real-world applications and hence have front-end components to handle them. The front-end components are typically optimized based on signal reconstruction objectives. This makes the overall speech processing system sub-optimal as the front-end is optimized independently of the downstream task. This approach also has another significant constraint that the enhancement/separation system can be trained with only simulated data and hence does not generalize well for real data. Alternatively, these front-end systems can be trained with application-oriented objectives. Emergent end-to-end neural methods have made it easier to optimize the frontend in such a manner. The goal of this work is to encompass carefully designed multichannel speech enhancement/separation subnetworks inside a sequence-to-sequence automatic speech recognition (ASR) system. This work takes an explainable AI approach to this problem where the intermediate outputs of the subnetworks can be interpreted although the entire network is trained only based on the speech recognition error minimization criteria. This proposal looks at two directions: (1) simultaneous dereverberation and denoising using a single differentiable speech recognition network which also learns some important hyperparameters from the data, (2) target speech extraction combining both anchor speech and location information which is optimized based on only the transcription as the target. In the first direction, dereverberation subnetwork is based on linear prediction where the filter order hyperparameter is estimated using a reinforcement learning approach, and the denoising (beamforming) subnetwork is based on a parametric multichannel Wiener filter where the speech distortion factor is also estimated inside the network. This method has shown a considerable gain in performance on real and unseen conditions. It is also shown how such a system optimized based on the ASR objective improves the speech enhancement quality on various signal level metrics in addition to the ASR word error rate (WER) metric. In the second direction, a location and anchor speech guided target speech extraction subnetwork is trained end-to-end with an ASR network. From experimental comparison with a traditional pipeline system, it is verified that this task can be realized by end-to-end ASR training objectives without using parallel clean data. The results are promising in mixtures of two speakers and noise. The future plan is to optimize an explicit source localization frontend with a speech recognition objective. This can play an important role in realizing a conversation system that recognizes who is speaking what, when, and where.

Apr
30
Thu
Thesis Proposal: Ke Li
Apr 30 @ 3:00 pm
Thesis Proposal: Ke Li

This presentation is happening remotely. Click this link as early as 15 minutes before the scheduled start time of the presentation to watch in a Zoom meeting.

Title: Context-aware Language Modeling and Adaptation for Automatic Speech Recognition

Abstract: Language models (LMs) are an important component in automatic speech recognition (ASR) and usually trained on transcriptions. Language use is strongly influenced by factors such as domain, topic, style, and user-preference. However, transcriptions from speech corpora are usually too limited to fully capture contextual variability in test domains. And some of the information is only available at test time. It is easily observed that the change of application domains often induces mismatch in lexicon and distribution of words. Even within the same domain, topics can shift and user-preference can vary. These observations indicate that LMs trained purely on transcriptions that may not be well representative for test domains are far from ideal and may severely affect ASR performance. To mitigate the mismatches, adapting LMs to contextual variables is desirable.

The goal of this work is to explore general and lightweight approaches for neural LM adaptation and context-aware modeling for ASR. In the adaptation direction, two approaches are investigated. The first is based on cache models. Although neural LMs outperform n-gram LMs on modeling longer context, previous studies show that some of them, for example, LSTMs, still only capture a relatively short span of context. Cache models that capture relatively long-term self-trigger information have been proved useful for n-gram LMs adaptation. This work extends a fast margin adaptation framework for neural LMs and adapts LSTM LMs in an unsupervised way. Specifically, pre-trained LMs are adapted to cache models estimated from decoded hypotheses. This method is lightweight as it does not require retraining. The second approach is interpolation-based. Linear interpolation is a simple and robust adaptation approach, while it is suboptimal since weights are globally optimized and not aware of local context. To tackle this issue, a mixer model that combines pre-trained neural LMs with dynamic weighting is proposed. Experimental results show that it outperforms finetuning and linear interpolation on most scenarios. As for context-aware modeling, this work proposes a simple and effective way to implicitly integrate cache models into neural LMs. It provides a simple alternative to the pointer sentinel mixture model. Experiments show that the proposed method is more effective on relatively rare words and outperforms several baselines. Future work is focused on analyzing the importance and the effect of various contextual factors on ASR and developing approaches for representing and modeling these factors to improve ASR performance.
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