The Nitech Text-To-Speech System for the Blizzard Challenge 2018
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Malware Classification with BERT
San Jose State University SJSU ScholarWorks Master's Projects Master's Theses and Graduate Research Spring 5-25-2021 Malware Classification with BERT Joel Lawrence Alvares Follow this and additional works at: https://scholarworks.sjsu.edu/etd_projects Part of the Artificial Intelligence and Robotics Commons, and the Information Security Commons Malware Classification with Word Embeddings Generated by BERT and Word2Vec Malware Classification with BERT Presented to Department of Computer Science San José State University In Partial Fulfillment of the Requirements for the Degree By Joel Alvares May 2021 Malware Classification with Word Embeddings Generated by BERT and Word2Vec The Designated Project Committee Approves the Project Titled Malware Classification with BERT by Joel Lawrence Alvares APPROVED FOR THE DEPARTMENT OF COMPUTER SCIENCE San Jose State University May 2021 Prof. Fabio Di Troia Department of Computer Science Prof. William Andreopoulos Department of Computer Science Prof. Katerina Potika Department of Computer Science 1 Malware Classification with Word Embeddings Generated by BERT and Word2Vec ABSTRACT Malware Classification is used to distinguish unique types of malware from each other. This project aims to carry out malware classification using word embeddings which are used in Natural Language Processing (NLP) to identify and evaluate the relationship between words of a sentence. Word embeddings generated by BERT and Word2Vec for malware samples to carry out multi-class classification. BERT is a transformer based pre- trained natural language processing (NLP) model which can be used for a wide range of tasks such as question answering, paraphrase generation and next sentence prediction. However, the attention mechanism of a pre-trained BERT model can also be used in malware classification by capturing information about relation between each opcode and every other opcode belonging to a malware family. -
Design of Optimal Feedback Filters with Guaranteed Closed-Loop Stability for Oversampled Noise-Shaping Subband Quantizers
49th IEEE Conference on Decision and Control December 15-17, 2010 Hilton Atlanta Hotel, Atlanta, GA, USA Design of Optimal Feedback Filters with Guaranteed Closed-Loop Stability for Oversampled Noise-Shaping Subband Quantizers SanderWahls HolgerBoche. Abstract—We consider the oversampled noise-shaping sub- T T T T ] ] ] ] band quantizer (ONSQ) with the quantization noise being q q v q,q v q,q + Quant- + + + + modeled as white noise. The problem at hand is to design an q - q izer q q - q q ,...,w ,...,w optimal feedback filter. Optimal feedback filters with regard 1 1 ,...,v ,...,v to the current standard model of the ONSQ inhabit several + q,1 q,1 w=[w + w=[w =[v =[v shortcomings: the stability of the feedback loop is not ensured - q q v v and the noise is considered independent of both the input and u ŋ u ŋ -1 -1 the feedback filter. Another previously unknown disadvantage, q z F(z) q q z F(z) q which we prove in our paper, is that optimal feedback filters in the standard model are never unique. The goal of this paper is (a) Real model: η is the qnt. error (b) Linearized model: η is white noise to show how these shortcomings can be overcome. The stability issue is addressed by showing that every feedback filter can be Figure 1: Noise-Shaping Quantizer “stabilized” in the sense that we can find another feedback filter which stabilizes the feedback loop and achieves a performance w v E (z) p 1 q,1 p R (z) equal (or arbitrarily close) to the original filter. -
ESE 531: Digital Signal Processing
ESE 531: Digital Signal Processing Lec 12: February 21st, 2017 Data Converters, Noise Shaping (con’t) Penn ESE 531 Spring 2017 - Khanna Lecture Outline ! Data Converters " Anti-aliasing " ADC " Quantization " Practical DAC ! Noise Shaping Penn ESE 531 Spring 2017 - Khanna 2 ADC Penn ESE 531 Spring 2017 - Khanna 3 Anti-Aliasing Filter with ADC Penn ESE 531 Spring 2017 - Khanna 4 Oversampled ADC Penn ESE 531 Spring 2017 - Khanna 5 Oversampled ADC Penn ESE 531 Spring 2017 - Khanna 6 Oversampled ADC Penn ESE 531 Spring 2017 - Khanna 7 Oversampled ADC Penn ESE 531 Spring 2017 - Khanna 8 Sampling and Quantization Penn ESE 531 Spring 2017 - Khanna 9 Sampling and Quantization Penn ESE 531 Spring 2017 - Khanna 10 Effect of Quantization Error on Signal ! Quantization error is a deterministic function of the signal " Consequently, the effect of quantization strongly depends on the signal itself ! Unless, we consider fairly trivial signals, a deterministic analysis is usually impractical " More common to look at errors from a statistical perspective " "Quantization noise” ! Two aspects " How much noise power (variance) does quantization add to our samples? " How is this noise distributed in frequency? Penn ESE 531 Spring 2017 - Khanna 11 Quantization Error ! Model quantization error as noise ! In that case: Penn ESE 531 Spring 2017 - Khanna 12 Ideal Quantizer ! Quantization step Δ ! Quantization error has sawtooth shape, ! Bounded by –Δ/2, +Δ/2 ! Ideally infinite input range and infinite number of quantization levels Penn ESE 568 Fall 2016 - Khanna adapted from Murmann EE315B, Stanford 13 Ideal B-bit Quantizer ! Practical quantizers have a limited input range and a finite set of output codes ! E.g. -
The Role of Higher-Level Linguistic Features in HMM-Based Speech Synthesis
INTERSPEECH 2010 The role of higher-level linguistic features in HMM-based speech synthesis Oliver Watts, Junichi Yamagishi, Simon King Centre for Speech Technology Research, University of Edinburgh, UK [email protected] [email protected] [email protected] Abstract the annotation of a synthesiser’s training data on listeners’ re- action to the speech produced by that synthesiser is still unclear We analyse the contribution of higher-level elements of the lin- to us. In particular, we suspect that features such as pitch ac- guistic specification of a data-driven speech synthesiser to the cent type which might be useful in voice-building if labelled naturalness of the synthetic speech which it generates. The reliably, are under-used in conventional systems because of la- system is trained using various subsets of the full feature-set, belling/prediction errors. For building the systems to be evalu- in which features relating to syntactic category, intonational ated we therefore used data for which hand labelled annotation phrase boundary, pitch accent and boundary tones are selec- of ToBI events is available. This allows us to compare ideal sys- tively removed. Utterances synthesised by the different config- tems built from a corpus where these higher-level features are urations of the system are then compared in a subjective evalu- accurately annotated with more conventional systems that rely ation of their naturalness. exclusively on prediction of these features from text for annota- The work presented forms background analysis for an on- tion of their training data. going set of experiments in performing text-to-speech (TTS) conversion based on shallow features: features that can be triv- 2. -
Learned in Speech Recognition: Contextual Acoustic Word Embeddings
LEARNED IN SPEECH RECOGNITION: CONTEXTUAL ACOUSTIC WORD EMBEDDINGS Shruti Palaskar∗, Vikas Raunak∗ and Florian Metze Carnegie Mellon University, Pittsburgh, PA, U.S.A. fspalaska j vraunak j fmetze [email protected] ABSTRACT model [10, 11, 12] trained for direct Acoustic-to-Word (A2W) speech recognition [13]. Using this model, we jointly learn to End-to-end acoustic-to-word speech recognition models have re- automatically segment and classify input speech into individual cently gained popularity because they are easy to train, scale well to words, hence getting rid of the problem of chunking or requiring large amounts of training data, and do not require a lexicon. In addi- pre-defined word boundaries. As our A2W model is trained at the tion, word models may also be easier to integrate with downstream utterance level, we show that we can not only learn acoustic word tasks such as spoken language understanding, because inference embeddings, but also learn them in the proper context of their con- (search) is much simplified compared to phoneme, character or any taining sentence. We also evaluate our contextual acoustic word other sort of sub-word units. In this paper, we describe methods embeddings on a spoken language understanding task, demonstrat- to construct contextual acoustic word embeddings directly from a ing that they can be useful in non-transcription downstream tasks. supervised sequence-to-sequence acoustic-to-word speech recog- Our main contributions in this paper are the following: nition model using the learned attention distribution. On a suite 1. We demonstrate the usability of attention not only for aligning of 16 standard sentence evaluation tasks, our embeddings show words to acoustic frames without any forced alignment but also for competitive performance against a word2vec model trained on the constructing Contextual Acoustic Word Embeddings (CAWE). -
Lecture 26 Word Embeddings and Recurrent Nets
CS447: Natural Language Processing http://courses.engr.illinois.edu/cs447 Where we’re at Lecture 25: Word Embeddings and neural LMs Lecture 26: Recurrent networks Lecture 26 Lecture 27: Sequence labeling and Seq2Seq Lecture 28: Review for the final exam Word Embeddings and Lecture 29: In-class final exam Recurrent Nets Julia Hockenmaier [email protected] 3324 Siebel Center CS447: Natural Language Processing (J. Hockenmaier) !2 What are neural nets? Simplest variant: single-layer feedforward net For binary Output unit: scalar y classification tasks: Single output unit Input layer: vector x Return 1 if y > 0.5 Recap Return 0 otherwise For multiclass Output layer: vector y classification tasks: K output units (a vector) Input layer: vector x Each output unit " yi = class i Return argmaxi(yi) CS447: Natural Language Processing (J. Hockenmaier) !3 CS447: Natural Language Processing (J. Hockenmaier) !4 Multi-layer feedforward networks Multiclass models: softmax(yi) We can generalize this to multi-layer feedforward nets Multiclass classification = predict one of K classes. Return the class i with the highest score: argmaxi(yi) Output layer: vector y In neural networks, this is typically done by using the softmax N Hidden layer: vector hn function, which maps real-valued vectors in R into a distribution … … … over the N outputs … … … … … …. For a vector z = (z0…zK): P(i) = softmax(zi) = exp(zi) ∕ ∑k=0..K exp(zk) Hidden layer: vector h1 (NB: This is just logistic regression) Input layer: vector x CS447: Natural Language Processing (J. Hockenmaier) !5 CS447: Natural Language Processing (J. Hockenmaier) !6 Neural Language Models LMs define a distribution over strings: P(w1….wk) LMs factor P(w1….wk) into the probability of each word: " P(w1….wk) = P(w1)·P(w2|w1)·P(w3|w1w2)·…· P(wk | w1….wk#1) A neural LM needs to define a distribution over the V words in Neural Language the vocabulary, conditioned on the preceding words. -
Self-Training Wavenet for TTS in Low-Data Regimes
StrawNet: Self-Training WaveNet for TTS in Low-Data Regimes Manish Sharma, Tom Kenter, Rob Clark Google UK fskmanish, tomkenter, [email protected] Abstract is increased. However, it can be seen from their results that the quality degrades when the number of recordings is further Recently, WaveNet has become a popular choice of neural net- decreased. work to synthesize speech audio. Autoregressive WaveNet is To reduce the voice artefacts observed in WaveNet stu- capable of producing high-fidelity audio, but is too slow for dent models trained under a low-data regime, we aim to lever- real-time synthesis. As a remedy, Parallel WaveNet was pro- age both the high-fidelity audio produced by an autoregressive posed, which can produce audio faster than real time through WaveNet, and the faster-than-real-time synthesis capability of distillation of an autoregressive teacher into a feedforward stu- a Parallel WaveNet. We propose a training paradigm, called dent network. A shortcoming of this approach, however, is that StrawNet, which stands for “Self-Training WaveNet”. The key a large amount of recorded speech data is required to produce contribution lies in using high-fidelity speech samples produced high-quality student models, and this data is not always avail- by an autoregressive WaveNet to self-train first a new autore- able. In this paper, we propose StrawNet: a self-training ap- gressive WaveNet and then a Parallel WaveNet model. We refer proach to train a Parallel WaveNet. Self-training is performed to models distilled this way as StrawNet student models. using the synthetic examples generated by the autoregressive We evaluate StrawNet by comparing it to a baseline WaveNet teacher. -
The RACAI Text-To-Speech Synthesis System
The RACAI Text-to-Speech Synthesis System Tiberiu Boroș, Radu Ion, Ștefan Daniel Dumitrescu Research Institute for Artificial Intelligence “Mihai Drăgănescu”, Romanian Academy (RACAI) [email protected], [email protected], [email protected] of standalone Natural Language Processing (NLP) Tools Abstract aimed at enabling text-to-speech (TTS) synthesis for less- This paper describes the RACAI Text-to-Speech (TTS) entry resourced languages. A good example is the case of for the Blizzard Challenge 2013. The development of the Romanian, a language which poses a lot of challenges for TTS RACAI TTS started during the Metanet4U project and the synthesis mainly because of its rich morphology and its system is currently part of the METASHARE platform. This reduced support in terms of freely available resources. Initially paper describes the work carried out for preparing the RACAI all the tools were standalone, but their design allowed their entry during the Blizzard Challenge 2013 and provides a integration into a single package and by adding a unit selection detailed description of our system and future development speech synthesis module based on the Pitch Synchronous directions. Overlap-Add (PSOLA) algorithm we were able to create a fully independent text-to-speech synthesis system that we refer Index Terms: speech synthesis, unit selection, concatenative to as RACAI TTS. 1. Introduction A considerable progress has been made since the start of the project, but our system still requires development and Text-to-speech (TTS) synthesis is a complex process that although considered premature, our participation in the addresses the task of converting arbitrary text into voice. -
Unsupervised Speech Representation Learning Using Wavenet Autoencoders Jan Chorowski, Ron J
1 Unsupervised speech representation learning using WaveNet autoencoders Jan Chorowski, Ron J. Weiss, Samy Bengio, Aaron¨ van den Oord Abstract—We consider the task of unsupervised extraction speaker gender and identity, from phonetic content, properties of meaningful latent representations of speech by applying which are consistent with internal representations learned autoencoding neural networks to speech waveforms. The goal by speech recognizers [13], [14]. Such representations are is to learn a representation able to capture high level semantic content from the signal, e.g. phoneme identities, while being desired in several tasks, such as low resource automatic speech invariant to confounding low level details in the signal such as recognition (ASR), where only a small amount of labeled the underlying pitch contour or background noise. Since the training data is available. In such scenario, limited amounts learned representation is tuned to contain only phonetic content, of data may be sufficient to learn an acoustic model on the we resort to using a high capacity WaveNet decoder to infer representation discovered without supervision, but insufficient information discarded by the encoder from previous samples. Moreover, the behavior of autoencoder models depends on the to learn the acoustic model and a data representation in a fully kind of constraint that is applied to the latent representation. supervised manner [15], [16]. We compare three variants: a simple dimensionality reduction We focus on representations learned with autoencoders bottleneck, a Gaussian Variational Autoencoder (VAE), and a applied to raw waveforms and spectrogram features and discrete Vector Quantized VAE (VQ-VAE). We analyze the quality investigate the quality of learned representations on LibriSpeech of learned representations in terms of speaker independence, the ability to predict phonetic content, and the ability to accurately re- [17]. -
Unsupervised Speech Representation Learning Using Wavenet Autoencoders
Unsupervised speech representation learning using WaveNet autoencoders https://arxiv.org/abs/1901.08810 Jan Chorowski University of Wrocław 06.06.2019 Deep Model = Hierarchy of Concepts Cat Dog … Moon Banana M. Zieler, “Visualizing and Understanding Convolutional Networks” Deep Learning history: 2006 2006: Stacked RBMs Hinton, Salakhutdinov, “Reducing the Dimensionality of Data with Neural Networks” Deep Learning history: 2012 2012: Alexnet SOTA on Imagenet Fully supervised training Deep Learning Recipe 1. Get a massive, labeled dataset 퐷 = {(푥, 푦)}: – Comp. vision: Imagenet, 1M images – Machine translation: EuroParlamanet data, CommonCrawl, several million sent. pairs – Speech recognition: 1000h (LibriSpeech), 12000h (Google Voice Search) – Question answering: SQuAD, 150k questions with human answers – … 2. Train model to maximize log 푝(푦|푥) Value of Labeled Data • Labeled data is crucial for deep learning • But labels carry little information: – Example: An ImageNet model has 30M weights, but ImageNet is about 1M images from 1000 classes Labels: 1M * 10bit = 10Mbits Raw data: (128 x 128 images): ca 500 Gbits! Value of Unlabeled Data “The brain has about 1014 synapses and we only live for about 109 seconds. So we have a lot more parameters than data. This motivates the idea that we must do a lot of unsupervised learning since the perceptual input (including proprioception) is the only place we can get 105 dimensions of constraint per second.” Geoff Hinton https://www.reddit.com/r/MachineLearning/comments/2lmo0l/ama_geoffrey_hinton/ Unsupervised learning recipe 1. Get a massive labeled dataset 퐷 = 푥 Easy, unlabeled data is nearly free 2. Train model to…??? What is the task? What is the loss function? Unsupervised learning by modeling data distribution Train the model to minimize − log 푝(푥) E.g. -
Real-Time Black-Box Modelling with Recurrent Neural Networks
Proceedings of the 22nd International Conference on Digital Audio Effects (DAFx-19), Birmingham, UK, September 2–6, 2019 REAL-TIME BLACK-BOX MODELLING WITH RECURRENT NEURAL NETWORKS Alec Wright, Eero-Pekka Damskägg, and Vesa Välimäki∗ Acoustics Lab, Department of Signal Processing and Acoustics Aalto University Espoo, Finland [email protected] ABSTRACT tube amplifiers and distortion pedals. In [14] it was shown that the WaveNet model of several distortion effects was capable of This paper proposes to use a recurrent neural network for black- running in real time. The resulting deep neural network model, box modelling of nonlinear audio systems, such as tube amplifiers however, was still fairly computationally expensive to run. and distortion pedals. As a recurrent unit structure, we test both Long Short-Term Memory and a Gated Recurrent Unit. We com- In this paper, we propose an alternative black-box model based pare the proposed neural network with a WaveNet-style deep neu- on an RNN. We demonstrate that the trained RNN model is capa- ral network, which has been suggested previously for tube ampli- ble of achieving the accuracy of the WaveNet model, whilst re- fier modelling. The neural networks are trained with several min- quiring considerably less processing power to run. The proposed utes of guitar and bass recordings, which have been passed through neural network, which consists of a single recurrent layer and a the devices to be modelled. A real-time audio plugin implement- fully connected layer, is suitable for real-time emulation of tube ing the proposed networks has been developed in the JUCE frame- amplifiers and distortion pedals. -
Synthesis and Recognition of Speech Creating and Listening to Speech
ISSN 1883-1974 (Print) ISSN 1884-0787 (Online) National Institute of Informatics News NII Interview 51 A Combination of Speech Synthesis and Speech Oct. 2014 Recognition Creates an Affluent Society NII Special 1 “Statistical Speech Synthesis” Technology with a Rapidly Growing Application Area NII Special 2 Finding Practical Application for Speech Recognition Feature Synthesis and Recognition of Speech Creating and Listening to Speech A digital book version of “NII Today” is now available. http://www.nii.ac.jp/about/publication/today/ This English language edition NII Today corresponds to No. 65 of the Japanese edition [Advance Notice] Great news! NII Interview Yamagishi-sensei will create my voice! Yamagishi One result is a speech translation sys- sound. Bit (NII Character) A Combination of Speech Synthesis tem. This system recognizes speech and translates it Ohkawara I would like your comments on the fu- using machine translation to synthesize speech, also ture challenges. and Speech Recognition automatically translating it into every language to Ono The challenge for speech recognition is how speak. Moreover, the speech is created with a human close it will come to humans in the distant speech A Word from the Interviewer voice. In second language learning, you can under- case. If this study is advanced, it will be possible to Creates an Affluent Society stand how you should pronounce it with your own summarize the contents of a meeting and to automati- voice. If this is further advanced, the system could cally take the minutes. If a robot understands the con- have an actor in a movie speak in a different language tents of conversations by multiple people in a natural More and more people have begun to use smart- a smartphone, I use speech input more often.