10. Pitch and Voicing Determination of Speech with an Extension Toward Music Signals Pitch Andw
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23 Strange Answers
23 ♦ Strange Answers Lawyer Plum was there and one pair of heirs when Otis Amber danced into the game room. “He-he-he, the Turtle’s lost its tail, I see.” Turtle slumped low in her chair. Flora Baumbach thought the short, sleek haircut was adorable, especially the way it swept forward over her little chin, but Turtle did not want to look adorable. She wanted to look mean. The dressmaker fumbled past the wad of money in her handbag. “Here, Alice, I thought you might like to see this.” Turtle glanced at the old snapshot. It’s Baba, all right, except younger. Same dumb smile. Suddenly she sat upright. “That’s my daughter, Rosalie,” Flora Baumbach said. “She must have been nine or ten when that picture was taken.” Rosalie was squat and square and squinty, her protruding tongue was too large for her mouth, her head lolled to one side. “I think I would have liked her, Baba,” Turtle said. “Rosalie looks like she was a very happy person. She must have been nice to have around.” Thump-thump, thump-thump. “Here come the victims,” Sydelle Pulaski announced. Angela greeted her sister with a wave of her crimson- streaked, healing hand. Turtle had convinced her not to confess: It would mean a criminal record, it would kill their mother, and no one would believe her anyhow. “I like your haircut.” “Thanks,” Turtle replied. Now Angela had to love her forever. Most of the heirs had to comment on Turtle’s hair. “You look like a real businesswoman,” Sandy said. -
Life on the Reef
INTERVIEW HOW TO BE… UK BAT DR SANDY KNAPP ON HER AN EXPERT SPECIES FASCINATION WITH PLANTS WITNESS 18 EXPLAINED THE SOCIETY OF BIOLOGY MAGAZINE ⁄www.societyofbiology.org ISSN 0006-3347 • Vol 62 No 3 • Jun/Jul 2015 LIFE ON THE REEF Unravelling the mysteries of coral NEWNEWNEW FROMFROM FROM GARLANDGARLAND GARLAND SCIENCE SCIENCE SCIENCE CellCellCell MembranesMembranes Membranes NEW FROM GARLANDLukasLukasLukas K.K. K. Buehler,Buehler, Buehler, SouthwesternSouthwestern Southwestern College, College, College, USA USASCIENCE USA CellCell CellMembranes Membranes Membranes offers offers offers a solida solid a solidfoundation foundation foundation for for for NEW FROM GARLANDunderstandingunderstandingunderstanding the the structurethe structure structure andSCIENCE and functionand function function of biologicalof ofbiological biological NEW FROM GARLANDmembranes.membranes.membranes. SCIENCE The book explores the composition and dynamics of cell TheThe book book explores explores the the composition composition and and dynamics dynamics of cellof cell membranes—discussing the molecular and biological NEW FROM GARLANDmembranes—discussingmembranes—discussing theSCIENCE the molecular molecular and and biological biological diversity of its lipid and protein components and how the diversitydiversity of itsof itslipid lipid and and protein protein components components and and how how the the combinatorialcombinatorial richness richness of both of bothcomponents components explains explains the the Cell Membranes chemical,combinatorial mechanical, -
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. -
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. -
Designing Speech and Multimodal Interactions for Mobile, Wearable, and Pervasive Applications
Designing Speech and Multimodal Interactions for Mobile, Wearable, and Pervasive Applications Cosmin Munteanu Nikhil Sharma Abstract University of Toronto Google, Inc. Traditional interfaces are continuously being replaced Mississauga [email protected] by mobile, wearable, or pervasive interfaces. Yet when [email protected] Frank Rudzicz it comes to the input and output modalities enabling Pourang Irani Toronto Rehabilitation Institute our interactions, we have yet to fully embrace some of University of Manitoba University of Toronto the most natural forms of communication and [email protected] [email protected] information processing that humans possess: speech, Sharon Oviatt Randy Gomez language, gestures, thoughts. Very little HCI attention Incaa Designs Honda Research Institute has been dedicated to designing and developing spoken [email protected] [email protected] language and multimodal interaction techniques, Matthew Aylett Keisuke Nakamura especially for mobile and wearable devices. In addition CereProc Honda Research Institute to the enormous, recent engineering progress in [email protected] [email protected] processing such modalities, there is now sufficient Gerald Penn Kazuhiro Nakadai evidence that many real-life applications do not require University of Toronto Honda Research Institute 100% accuracy of processing multimodal input to be [email protected] [email protected] useful, particularly if such modalities complement each Shimei Pan other. This multidisciplinary, two-day workshop -
Introduction to Digital Speech Processing Schafer Rabiner and Ronald W
SIGv1n1-2.qxd 11/20/2007 3:02 PM Page 1 FnT SIG 1:1-2 Foundations and Trends® in Signal Processing Introduction to Digital Speech Processing Processing to Digital Speech Introduction R. Lawrence W. Rabiner and Ronald Schafer 1:1-2 (2007) Lawrence R. Rabiner and Ronald W. Schafer Introduction to Digital Speech Processing highlights the central role of DSP techniques in modern speech communication research and applications. It presents a comprehensive overview of digital speech processing that ranges from the basic nature of the speech signal, through a variety of methods of representing speech in digital form, to applications in voice communication and automatic synthesis and recognition of speech. Introduction to Digital Introduction to Digital Speech Processing provides the reader with a practical introduction to Speech Processing the wide range of important concepts that comprise the field of digital speech processing. It serves as an invaluable reference for students embarking on speech research as well as the experienced researcher already working in the field, who can utilize the book as a Lawrence R. Rabiner and Ronald W. Schafer reference guide. This book is originally published as Foundations and Trends® in Signal Processing Volume 1 Issue 1-2 (2007), ISSN: 1932-8346. now now the essence of knowledge Introduction to Digital Speech Processing Introduction to Digital Speech Processing Lawrence R. Rabiner Rutgers University and University of California Santa Barbara USA [email protected] Ronald W. Schafer Hewlett-Packard Laboratories Palo Alto, CA USA Boston – Delft Foundations and Trends R in Signal Processing Published, sold and distributed by: now Publishers Inc. -
Indie TV: Innovation in Series Development Aymar Jean Christian Northwestern Univeristy I Spent Five Years Interviewing Creators
Indie TV: Innovation in series development DRAFT Aymar Jean Christian Northwestern Univeristy I spent five years interviewing creators of American web series, or independent television online,1 speaking with 134 producers and executives making and releasing scripted series independent of traditional network investment. No two series or creators were ever alike. Some were sci-fi comedies created by Trekkers, others urban dramas by black women. Nevertheless nearly every creator differentiated their work and products from traditional television in some way. Strikingly, whenever I ask if any particular creator inspired them to develop their own show independently, I heard one response more than any other: Felicia Day. Indie TV creators revere Day for her innovation in production and distribution: she developed a hit franchise, The Guild, outside network development processes. The Guild, a sitcom about a diverse group of gamers who sit at computers all day to play an MMORPG in the style of World of Warcraft, ran for six seasons, from 2007 to 2013, and spawned a music video, DVDs, comic books, a companion book, as well as marshaled the passions of hundreds of thousands of fans. By the end of its run, The Guild had amassed over 330,000 subscribers on YouTube, though it could also be seen on other portals. In my work I have seen The Guild emerge as central to the online indie TV industry’s case for itself as antidote to what ails Hollywood. Day initially pitched to Christian, A.J. (forthcoming). Indie TV: Innovation in Series Development, in Media Independence: 1 working with freedom or working for free?. -
Models of Speech Synthesis ROLF CARLSON Department of Speech Communication and Music Acoustics, Royal Institute of Technology, S-100 44 Stockholm, Sweden
Proc. Natl. Acad. Sci. USA Vol. 92, pp. 9932-9937, October 1995 Colloquium Paper This paper was presented at a colloquium entitled "Human-Machine Communication by Voice," organized by Lawrence R. Rabiner, held by the National Academy of Sciences at The Arnold and Mabel Beckman Center in Irvine, CA, February 8-9,1993. Models of speech synthesis ROLF CARLSON Department of Speech Communication and Music Acoustics, Royal Institute of Technology, S-100 44 Stockholm, Sweden ABSTRACT The term "speech synthesis" has been used need large amounts of speech data. Models working close to for diverse technical approaches. In this paper, some of the the waveform are now typically making use of increased unit approaches used to generate synthetic speech in a text-to- sizes while still modeling prosody by rule. In the middle of the speech system are reviewed, and some of the basic motivations scale, "formant synthesis" is moving toward the articulatory for choosing one method over another are discussed. It is models by looking for "higher-level parameters" or to larger important to keep in mind, however, that speech synthesis prestored units. Articulatory synthesis, hampered by lack of models are needed not just for speech generation but to help data, still has some way to go but is yielding improved quality, us understand how speech is created, or even how articulation due mostly to advanced analysis-synthesis techniques. can explain language structure. General issues such as the synthesis of different voices, accents, and multiple languages Flexibility and Technical Dimensions are discussed as special challenges facing the speech synthesis community. -
Audio Processing Laboratory
AC 2010-1594: A GRADUATE LEVEL COURSE: AUDIO PROCESSING LABORATORY Buket Barkana, University of Bridgeport Page 15.35.1 Page © American Society for Engineering Education, 2010 A Graduate Level Course: Audio Processing Laboratory Abstract Audio signal processing is a part of the digital signal processing (DSP) field in science and engineering that has developed rapidly over the past years. Expertise in audio signal processing - including speech signal processing- is becoming increasingly important for working engineers from diverse disciplines. Audio signals are stored, processed, and transmitted using digital techniques. Creating these technologies requires engineers that understand real-time application issues in the related areas. With this motivation, we designed a graduate level laboratory course which is Audio Processing Laboratory in the electrical engineering department in our school two years ago. This paper presents the status of the audio processing laboratory course in our school. We report the challenges that we have faced during the development of this course in our school and discuss the instructor’s and students’ assessments and recommendations in this real-time signal-processing laboratory course. 1. Introduction Many DSP laboratory courses are developed for undergraduate and graduate level engineering education. These courses mostly focus on the general signal processing techniques such as quantization, filter design, Fast Fourier Transformation (FFT), and spectral analysis [1-3]. Because of the increasing popularity of Web-based education and the advancements in streaming media applications, several web-based DSP Laboratory courses have been designed for distance education [4-8]. An internet-based signal processing laboratory that provides hands-on learning experiences in distributed learning environments has been developed by Spanias et.al[6] . -
ECE438 - Digital Signal Processing with Applications 1
Purdue University: ECE438 - Digital Signal Processing with Applications 1 ECE438 - Laboratory 9: Speech Processing (Week 1) October 6, 2010 1 Introduction Speech is an acoustic waveform that conveys information from a speaker to a listener. Given the importance of this form of communication, it is no surprise that many applications of signal processing have been developed to manipulate speech signals. Almost all speech processing applications currently fall into three broad categories: speech recognition, speech synthesis, and speech coding. Speech recognition may be concerned with the identification of certain words, or with the identification of the speaker. Isolated word recognition algorithms attempt to identify individual words, such as in automated telephone services. Automatic speech recognition systems attempt to recognize continuous spoken language, possibly to convert into text within a word processor. These systems often incorporate grammatical cues to increase their accuracy. Speaker identification is mostly used in security applications, as a person’s voice is much like a “fingerprint”. The objective in speech synthesis is to convert a string of text, or a sequence of words, into natural-sounding speech. One example is the Speech Plus synthesizer used by Stephen Hawking (although it unfortunately gives him an American accent). There are also similar systems which read text for the blind. Speech synthesis has also been used to aid scientists in learning about the mechanisms of human speech production, and thereby in the treatment of speech-related disorders. Speech coding is mainly concerned with exploiting certain redundancies of the speech signal, allowing it to be represented in a compressed form. Much of the research in speech compression has been motivated by the need to conserve bandwidth in communication sys- tems. -
Speech Processing Based on a Sinusoidal Model
R.J. McAulay andT.F. Quatieri Speech Processing Based on a Sinusoidal Model Using a sinusoidal model of speech, an analysis/synthesis technique has been devel oped that characterizes speech in terms of the amplitudes, frequencies, and phases of the component sine waves. These parameters can be estimated by applying a simple peak-picking algorithm to a short-time Fourier transform (STFf) of the input speech. Rapid changes in the highly resolved spectral components are tracked by using a frequency-matching algorithm and the concept of"birth" and "death" ofthe underlying sinewaves. Fora givenfrequency track, a cubic phasefunction is applied to a sine-wave generator. whose outputis amplitude-modulated andadded to the sine waves generated for the other frequency tracks. and this sum is the synthetic speech output. The resulting waveform preserves the general waveform shape and is essentially indistin gUishable from the original speech. Furthermore. inthe presence ofnoise the perceptual characteristics ofthe speech and the noise are maintained. Itwas also found that high quality reproduction was obtained for a large class of inputs: two overlapping. super posed speech waveforms; music waveforms; speech in musical backgrounds; and certain marine biologic sounds. The analysis/synthesis system has become the basis for new approaches to such diverse applications as multirate coding for secure communications, time-scale and pitch-scale algorithms for speech transformations, and phase dispersion for the enhancement ofAM radio broadcasts. Moreover, the technology behind the applications has been successfully transferred to private industry for commercial development. Speech signals can be represented with a Other approaches to analysis/synthesis that· speech production model that views speech as are based on sine-wave models have been the result of passing a glottal excitation wave discussed. -
Edinburgh Research Explorer
Edinburgh Research Explorer Designing Speech and Multimodal Interactions for Mobile, Wearable, and Pervasive Applications Citation for published version: Munteanu, C, Irani, P, Oviatt, S, Aylett, M, Penn, G, Pan, S, Sharma, N, Rudzicz, F, Gomez, R, Nakamura, K & Nakadai, K 2016, Designing Speech and Multimodal Interactions for Mobile, Wearable, and Pervasive Applications. in Proceedings of the 2016 CHI Conference Extended Abstracts on Human Factors in Computing Systems. CHI EA '16, ACM, New York, NY, USA, pp. 3612-3619, 2016 CHI Conference Extended Abstracts on Human Factors in Computing Systems, San Jose, United States, 7/05/16. https://doi.org/10.1145/2851581.2856506 Digital Object Identifier (DOI): 10.1145/2851581.2856506 Link: Link to publication record in Edinburgh Research Explorer Document Version: Peer reviewed version Published In: Proceedings of the 2016 CHI Conference Extended Abstracts on Human Factors in Computing Systems General rights Copyright for the publications made accessible via the Edinburgh Research Explorer is retained by the author(s) and / or other copyright owners and it is a condition of accessing these publications that users recognise and abide by the legal requirements associated with these rights. Take down policy The University of Edinburgh has made every reasonable effort to ensure that Edinburgh Research Explorer content complies with UK legislation. If you believe that the public display of this file breaches copyright please contact [email protected] providing details, and we will remove access to the work immediately and investigate your claim. Download date: 28. Sep. 2021 Designing Speech and Multimodal Interactions for Mobile, Wearable, and Pervasive Applications Cosmin Munteanu Nikhil Sharma University of Toronto Google, Inc.