Deep Learning for Audio Signal Processing
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Automatic Feature Learning Using Recurrent Neural Networks
Predicting purchasing intent: Automatic Feature Learning using Recurrent Neural Networks Humphrey Sheil Omer Rana Ronan Reilly Cardiff University Cardiff University Maynooth University Cardiff, Wales Cardiff, Wales Maynooth, Ireland [email protected] [email protected] [email protected] ABSTRACT influence three out of four major variables that affect profit. Inaddi- We present a neural network for predicting purchasing intent in an tion, merchants increasingly rely on (and pay advertising to) much Ecommerce setting. Our main contribution is to address the signifi- larger third-party portals (for example eBay, Google, Bing, Taobao, cant investment in feature engineering that is usually associated Amazon) to achieve their distribution, so any direct measures the with state-of-the-art methods such as Gradient Boosted Machines. merchant group can use to increase their profit is sorely needed. We use trainable vector spaces to model varied, semi-structured input data comprising categoricals, quantities and unique instances. McKinsey A.T. Kearney Affected by Multi-layer recurrent neural networks capture both session-local shopping intent and dataset-global event dependencies and relationships for user Price management 11.1% 8.2% Yes sessions of any length. An exploration of model design decisions Variable cost 7.8% 5.1% Yes including parameter sharing and skip connections further increase Sales volume 3.3% 3.0% Yes model accuracy. Results on benchmark datasets deliver classifica- Fixed cost 2.3% 2.0% No tion accuracy within 98% of state-of-the-art on one and exceed state-of-the-art on the second without the need for any domain / Table 1: Effect of improving different variables on operating dataset-specific feature engineering on both short and long event profit, from [22]. -
Exploring the Potential of Sparse Coding for Machine Learning
Portland State University PDXScholar Dissertations and Theses Dissertations and Theses 10-19-2020 Exploring the Potential of Sparse Coding for Machine Learning Sheng Yang Lundquist Portland State University Follow this and additional works at: https://pdxscholar.library.pdx.edu/open_access_etds Part of the Applied Mathematics Commons, and the Computer Sciences Commons Let us know how access to this document benefits ou.y Recommended Citation Lundquist, Sheng Yang, "Exploring the Potential of Sparse Coding for Machine Learning" (2020). Dissertations and Theses. Paper 5612. https://doi.org/10.15760/etd.7484 This Dissertation is brought to you for free and open access. It has been accepted for inclusion in Dissertations and Theses by an authorized administrator of PDXScholar. Please contact us if we can make this document more accessible: [email protected]. Exploring the Potential of Sparse Coding for Machine Learning by Sheng Y. Lundquist A dissertation submitted in partial fulfillment of the requirements for the degree of Doctor of Philosophy in Computer Science Dissertation Committee: Melanie Mitchell, Chair Feng Liu Bart Massey Garrett Kenyon Bruno Jedynak Portland State University 2020 © 2020 Sheng Y. Lundquist Abstract While deep learning has proven to be successful for various tasks in the field of computer vision, there are several limitations of deep-learning models when com- pared to human performance. Specifically, human vision is largely robust to noise and distortions, whereas deep learning performance tends to be brittle to modifi- cations of test images, including being susceptible to adversarial examples. Addi- tionally, deep-learning methods typically require very large collections of training examples for good performance on a task, whereas humans can learn to perform the same task with a much smaller number of training examples. -
And Passive Speakers?
To be active or not to be active – that is the question... To be active or not to be active – that is the question... 1. Active, passive – the situation 2. Active and passive loudspeaker – the basic difference 3. Passive loudspeaker 4. Active loudspeaker 5. The ADAM loudspeaker: passive option, active optimum 1. Active versus Passive – the Situation In any hifi-system, the loudspeakers are the pivotal component concerning sound quality. That is not to say that the other components do not matter. Nevertheless, it is indisputable that the loudspeaker is decisive for the sound of a hifi-system. It is – besides the acoustical properties of the listening room and the recording itself – the core of any music reproduction. The history of loudspeaker development has produced a great variety of very different systems and designs. The circuit technology of the frequency-separating filter that separates the audio signal into different frequency ranges is determining the design of a loudspeaker. In this respect we distinguish between active and passive systems. Usually, this is a topic that is often underestimated in its importance for sound quality. Active-passive is much more than just a technical negligibility: In fact, the impact of the dividing network on the overall sound of a loudspeaker is substantial. Active or passive – which system is preferable? Considering the aspects mentioned before, it may become a little more comprehensive why the very question comes up over and over again in the hifi-world. For decades it has been spooking as a debate on principles in the journals and magazines and for some time, now, in the web forums. -
Enhance Your DSP Course with These Interesting Projects
AC 2012-3836: ENHANCE YOUR DSP COURSE WITH THESE INTER- ESTING PROJECTS Dr. Joseph P. Hoffbeck, University of Portland Joseph P. Hoffbeck is an Associate Professor of electrical engineering at the University of Portland in Portland, Ore. He has a Ph.D. from Purdue University, West Lafayette, Ind. He previously worked with digital cell phone systems at Lucent Technologies (formerly AT&T Bell Labs) in Whippany, N.J. His technical interests include communication systems, digital signal processing, and remote sensing. Page 25.566.1 Page c American Society for Engineering Education, 2012 Enhance your DSP Course with these Interesting Projects Abstract Students are often more interested learning technical material if they can see useful applications for it, and in digital signal processing (DSP) it is possible to develop homework assignments, projects, or lab exercises to show how the techniques can be used in realistic situations. This paper presents six simple, yet interesting projects that are used in the author’s undergraduate digital signal processing course with the objective of motivating the students to learn how to use the Fast Fourier Transform (FFT) and how to design digital filters. Four of the projects are based on the FFT, including a simple voice recognition algorithm that determines if an audio recording contains “yes” or “no”, a program to decode dual-tone multi-frequency (DTMF) signals, a project to determine which note is played by a musical instrument and if it is sharp or flat, and a project to check the claim that cars honk in the tone of F. Two of the projects involve designing filters to eliminate noise from audio recordings, including designing a lowpass filter to remove a truck backup beeper from a recording of an owl hooting and designing a highpass filter to remove jet engine noise from a recording of birds chirping. -
Speech Signal Basics
Speech Signal Basics Nimrod Peleg Updated: Feb. 2010 Course Objectives • To get familiar with: – Speech coding in general – Speech coding for communication (military, cellular) • Means: – Introduction the basics of speech processing – Presenting an overview of speech production and hearing systems – Focusing on speech coding: LPC codecs: • Basic principles • Discussing of related standards • Listening and reviewing different codecs. הנחיות לשימוש נכון בטלפון, פלשתינה-א"י, 1925 What is Speech ??? • Meaning: Text, Identity, Punctuation, Emotion –prosody: rhythm, pitch, intensity • Statistics: sampled speech is a string of numbers – Distribution (Gamma, Laplace) Quantization • Information Theory: statistical redundancy – ZIP, ARJ, compress, Shorten, Entropy coding… ...1001100010100110010... • Perceptual redundancy – Temporal masking , frequency masking (mp3, Dolby) • Speech production Model – Physical system: Linear Prediction The Speech Signal • Created at the Vocal cords, Travels through the Vocal tract, and Produced at speakers mouth • Gets to the listeners ear as a pressure wave • Non-Stationary, but can be divided to sound segments which have some common acoustic properties for a short time interval • Two Major classes: Vowels and Consonants Speech Production • A sound source excites a (vocal tract) filter – Voiced: Periodic source, created by vocal cords – UnVoiced: Aperiodic and noisy source •The Pitch is the fundamental frequency of the vocal cords vibration (also called F0) followed by 4-5 Formants (F1 -F5) at higher frequencies. -
Text-To-Speech Synthesis
Generative Model-Based Text-to-Speech Synthesis Heiga Zen (Google London) February rd, @MIT Outline Generative TTS Generative acoustic models for parametric TTS Hidden Markov models (HMMs) Neural networks Beyond parametric TTS Learned features WaveNet End-to-end Conclusion & future topics Outline Generative TTS Generative acoustic models for parametric TTS Hidden Markov models (HMMs) Neural networks Beyond parametric TTS Learned features WaveNet End-to-end Conclusion & future topics Text-to-speech as sequence-to-sequence mapping Automatic speech recognition (ASR) “Hello my name is Heiga Zen” ! Machine translation (MT) “Hello my name is Heiga Zen” “Ich heiße Heiga Zen” ! Text-to-speech synthesis (TTS) “Hello my name is Heiga Zen” ! Heiga Zen Generative Model-Based Text-to-Speech Synthesis February rd, of Speech production process text (concept) fundamental freq voiced/unvoiced char freq transfer frequency speech transfer characteristics magnitude start--end Sound source fundamental voiced: pulse frequency unvoiced: noise modulation of carrier wave by speech information air flow Heiga Zen Generative Model-Based Text-to-Speech Synthesis February rd, of Typical ow of TTS system TEXT Sentence segmentation Word segmentation Text normalization Text analysis Part-of-speech tagging Pronunciation Speech synthesis Prosody prediction discrete discrete Waveform generation ) NLP Frontend discrete continuous SYNTHESIZED ) SEECH Speech Backend Heiga Zen Generative Model-Based Text-to-Speech Synthesis February rd, of Rule-based, formant synthesis [] -
Real-Time Programming and Processing of Music Signals Arshia Cont
Real-time Programming and Processing of Music Signals Arshia Cont To cite this version: Arshia Cont. Real-time Programming and Processing of Music Signals. Sound [cs.SD]. Université Pierre et Marie Curie - Paris VI, 2013. tel-00829771 HAL Id: tel-00829771 https://tel.archives-ouvertes.fr/tel-00829771 Submitted on 3 Jun 2013 HAL is a multi-disciplinary open access L’archive ouverte pluridisciplinaire HAL, est archive for the deposit and dissemination of sci- destinée au dépôt et à la diffusion de documents entific research documents, whether they are pub- scientifiques de niveau recherche, publiés ou non, lished or not. The documents may come from émanant des établissements d’enseignement et de teaching and research institutions in France or recherche français ou étrangers, des laboratoires abroad, or from public or private research centers. publics ou privés. Realtime Programming & Processing of Music Signals by ARSHIA CONT Ircam-CNRS-UPMC Mixed Research Unit MuTant Team-Project (INRIA) Musical Representations Team, Ircam-Centre Pompidou 1 Place Igor Stravinsky, 75004 Paris, France. Habilitation à diriger la recherche Defended on May 30th in front of the jury composed of: Gérard Berry Collège de France Professor Roger Dannanberg Carnegie Mellon University Professor Carlos Agon UPMC - Ircam Professor François Pachet Sony CSL Senior Researcher Miller Puckette UCSD Professor Marco Stroppa Composer ii à Marie le sel de ma vie iv CONTENTS 1. Introduction1 1.1. Synthetic Summary .................. 1 1.2. Publication List 2007-2012 ................ 3 1.3. Research Advising Summary ............... 5 2. Realtime Machine Listening7 2.1. Automatic Transcription................. 7 2.2. Automatic Alignment .................. 10 2.2.1. -
Introduction to the Digital Snake
TABLE OF CONTENTS What’s an Audio Snake ........................................4 The Benefits of the Digital Snake .........................5 Digital Snake Components ..................................6 Improved Intelligibility ...........................................8 Immunity from Hums & Buzzes .............................9 Lightweight & Portable .......................................10 Low Installation Cost ...........................................11 Additional Benefits ..............................................12 Digital Snake Comparison Chart .......................14 Conclusion ...........................................................15 All rights reserved. No part of this publication may be reproduced in any form without the written permission of Roland System Solutions. All trade- marks are the property of their respective owners. Roland System Solutions © 2005 Introduction Digital is the technology of our world today. It’s all around us in the form of CDs, DVDs, MP3 players, digital cameras, and computers. Digital offers great benefits to all of us, and makes our lives easier and better. Such benefits would have been impossible using analog technology. Who would go back to the world of cassette tapes, for example, after experiencing the ease of access and clean sound quality of a CD? Until recently, analog sound systems have been the standard for sound reinforcement and PA applications. However, recent technological advances have brought the benefits of digital audio to the live sound arena. Digital audio is superior -
Benchmarking Audio Signal Representation Techniques for Classification with Convolutional Neural Networks
sensors Article Benchmarking Audio Signal Representation Techniques for Classification with Convolutional Neural Networks Roneel V. Sharan * , Hao Xiong and Shlomo Berkovsky Australian Institute of Health Innovation, Macquarie University, Sydney, NSW 2109, Australia; [email protected] (H.X.); [email protected] (S.B.) * Correspondence: [email protected] Abstract: Audio signal classification finds various applications in detecting and monitoring health conditions in healthcare. Convolutional neural networks (CNN) have produced state-of-the-art results in image classification and are being increasingly used in other tasks, including signal classification. However, audio signal classification using CNN presents various challenges. In image classification tasks, raw images of equal dimensions can be used as a direct input to CNN. Raw time- domain signals, on the other hand, can be of varying dimensions. In addition, the temporal signal often has to be transformed to frequency-domain to reveal unique spectral characteristics, therefore requiring signal transformation. In this work, we overview and benchmark various audio signal representation techniques for classification using CNN, including approaches that deal with signals of different lengths and combine multiple representations to improve the classification accuracy. Hence, this work surfaces important empirical evidence that may guide future works deploying CNN for audio signal classification purposes. Citation: Sharan, R.V.; Xiong, H.; Berkovsky, S. Benchmarking Audio Keywords: convolutional neural networks; fusion; interpolation; machine learning; spectrogram; Signal Representation Techniques for time-frequency image Classification with Convolutional Neural Networks. Sensors 2021, 21, 3434. https://doi.org/10.3390/ s21103434 1. Introduction Sensing technologies find applications in detecting and monitoring health conditions. -
Audio Interface User's Manual
Audio Interface User’s Manual Eurorack Synthesizer Modules 14 HP TABLE OF CONTENTS 1.INTRODUCTION 2.WARRANTY 3.INSTALLATION 4.FUNCTION OF PANEL COMPONENTS 5.SIGNALFLOW & ROUTING 6.SPECIFICATIONS 2 1. INTRODUCTION Audiophile Circuits League. -The main purpose of the ACL Audio Interface module is to interface modular synthesizer systems with professional audio recording and stage equipment. The combination of studio quality signal path, �lexible routing possibilities and a headphoneheadphones ampli�ier, with low capabledistortion, of drivingmakes theboth connection high and low between impedance these different environments effortless and sonically transparent. The ACL Audio Interface offers balanced to unbalanced and unbalanced to balanced stereo lines with level controls. The stereo signal from the auxiliary input,either alsowith with balanced level tocontrol, unbalanced, can be oroptionally unbalanced routed to balanced to and mixed line signals,together or can be muted. The headphone ampli�ier can also get its signal from one or the other line after the level control and mixing stage, or can be muted. Since the ampli�ier is AC coupled only at the input, but not at the output, there is an on-board DC protection circuit included. In case the headphone ampli�ier is driven into clipping, the protection can also be tripped. The module has a soft start function* and one overload indicator for every line. *With the soft start function, the interface switch is turned on after a while after turning on the Eurorack main unit. This function can prevent output of unexpecteddamage to the sound speaker. that another module will emit at startup, which will cause 3 2. -
21065L Audio Tutorial
a Using The Low-Cost, High Performance ADSP-21065L Digital Signal Processor For Digital Audio Applications Revision 1.0 - 12/4/98 dB +12 0 -12 Left Right Left EQ Right EQ Pan L R L R L R L R L R L R L R L R 1 2 3 4 5 6 7 8 Mic High Line L R Mid Play Back Bass CNTR 0 0 3 4 Input Gain P F R Master Vol. 1 2 3 4 5 6 7 8 Authors: John Tomarakos Dan Ledger Analog Devices DSP Applications 1 Using The Low Cost, High Performance ADSP-21065L Digital Signal Processor For Digital Audio Applications Dan Ledger and John Tomarakos DSP Applications Group, Analog Devices, Norwood, MA 02062, USA This document examines desirable DSP features to consider for implementation of real time audio applications, and also offers programming techniques to create DSP algorithms found in today's professional and consumer audio equipment. Part One will begin with a discussion of important audio processor-specific characteristics such as speed, cost, data word length, floating-point vs. fixed-point arithmetic, double-precision vs. single-precision data, I/O capabilities, and dynamic range/SNR capabilities. Comparisions between DSP's and audio decoders that are targeted for consumer/professional audio applications will be shown. Part Two will cover example algorithmic building blocks that can be used to implement many DSP audio algorithms using the ADSP-21065L including: Basic audio signal manipulation, filtering/digital parametric equalization, digital audio effects and sound synthesis techniques. TABLE OF CONTENTS 0. INTRODUCTION ................................................................................................................................................................4 1. -
4 – Synthesis and Analysis of Complex Waves; Fourier Spectra
4 – Synthesis and Analysis of Complex Waves; Fourier spectra Complex waves Many physical systems (such as music instruments) allow existence of a particular set of standing waves with the frequency spectrum consisting of the fundamental (minimal allowed) frequency f1 and the overtones or harmonics fn that are multiples of f1, that is, fn = n f1, n=2,3,…The amplitudes of the harmonics An depend on how exactly the system is excited (plucking a guitar string at the middle or near an end). The sound emitted by the system and registered by our ears is thus a complex wave (or, more precisely, a complex oscillation ), or just signal, that has the general form nmax = π +ϕ = 1 W (t) ∑ An sin( 2 ntf n ), periodic with period T n=1 f1 Plots of W(t) that can look complicated are called wave forms . The maximal number of harmonics nmax is actually infinite but we can take a finite number of harmonics to synthesize a complex wave. ϕ The phases n influence the wave form a lot, visually, but the human ear is insensitive to them (the psychoacoustical Ohm‘s law). What the ear hears is the fundamental frequency (even if it is missing in the wave form, A1=0 !) and the amplitudes An that determine the sound quality or timbre . 1 Examples of synthesis of complex waves 1 1. Triangular wave: A = for n odd and zero for n even (plotted for f1=500) n n2 W W nmax =1, pure sinusoidal 1 nmax =3 1 0.5 0.5 t 0.001 0.002 0.003 0.004 t 0.001 0.002 0.003 0.004 £ 0.5 ¢ 0.5 £ 1 ¢ 1 W W n =11, cos ¤ sin 1 nmax =11 max 0.75 0.5 0.5 0.25 t 0.001 0.002 0.003 0.004 t 0.001 0.002 0.003 0.004 ¡ 0.5 0.25 0.5 ¡ 1 Acoustically equivalent 2 0.75 1 1.