@book {3452, title = {The Science of Sound Recording}, year = {2012}, pages = {256}, publisher = {Focal Press}, organization = {Focal Press}, edition = {1}, address = {Waltham, MA}, issn = {978-0-240-82154-2}, url = {http://www.focalpress.com/books/audio/the_science_of_sound_recording.aspx}, author = {Jay Kadis} } @inbook {4506, title = {Don{\textquoteright}t Forget the Laptop: Using Native Input Capabilities for Expressive Musical Control}, booktitle = {A NIME Reader: Fifteen Years of New Interfaces for Musical Expression}, year = {2017}, note = {\ }, publisher = {Springer}, organization = {Springer}, abstract = {\ }, keywords = {aesthetics, design, instrument design, laptop orchestra}, issn = {978-3-319-47214-0}, url = {http://smelt.cs.princeton.edu/}, author = {Rebecca Fiebrink and Ge Wang and Perry R. Cook}, editor = {Alexander Refsum Jensenius and Michael Lyons} } @inbook {2585, title = {A History of Programming and Music}, booktitle = {Cambridge Companion to Electronic Music}, year = {2008}, publisher = {Cambridge University Press}, organization = {Cambridge University Press}, abstract = {

Computer music now has a practical history of more than fifty years,\ though Ada Lovelace had anticipated applications in 1843. An involvement\ of computers in music, with their potential for fantastical automations,\ necessarily saw the use of programming languages. In the present day, a\ bewildering array of languages exist, some specially developed with\ issues of sound and music representation in mind. Whilst entry-level\ softwares exist to hide complicated programming from end-users, the\ musician seeking custom abstractions and intimate control of all\ parameters is drawn to programming interfaces. This chapter provides a\ historical perspective on computer music programming, from Music N to\ Csound, from Max to Max/MSP, and details many interesting projects such\ as Common Lisp Music, Nyquist, SuperCollider, and ChucK. Interesting\ applications from different eras are highlighted including Hiller and\ Isaacson{\textquoteright}s Illiac Suite, the Computer Audio Research Laboratory (CARL),\ the Composer{\textquoteright}s Desktop Project, writing custom sound software, plug-ins\ and extensions, and live coding.

}, keywords = {computer music, history, language, programming}, author = {Ge Wang} } @inbook {2799, title = {Introduction to Digital Filter Theory}, booktitle = {Digital Audio Signal Processing: An Anthology}, year = {1985}, month = {04/1985}, publisher = {William Kaufmann}, organization = {William Kaufmann}, address = {Los Altos, CA}, keywords = {digital signal processing, dsp, filter theory}, url = {https://ccrma.stanford.edu/files/papers/stanm20.pdf}, author = {Julius O. Smith}, editor = {Strawn, J.} } @inbook {2766, title = {An Introduction to the Phase Vocoder}, booktitle = {Digital Audio Signal Processing: An Anthology}, year = {1975}, month = {02/1987}, publisher = {William Kaufmann}, organization = {William Kaufmann}, address = {Los Altos, CA}, keywords = {phase vocoder}, url = {https://ccrma.stanford.edu/files/papers/stanm55.pdf}, author = {John W. Gordon and Strawn, J.}, editor = {Strawn, J.} } @inbook {4493, title = {Live Auralization of Cappella Romana at the Bing Concert Hall, Stanford University}, booktitle = {Aural Architecture in Byzantium: Music, Acoustics, and Ritual}, year = {2017}, note = {\ }, publisher = {Routledge}, organization = {Routledge}, abstract = {\ }, author = {Abel, Jonathan S and Kurt James Werner}, editor = {Pentcheva, Bissera V.} } @inbook {2591, title = {The Mobile Phone Orchestra}, booktitle = {Oxford Handbook of Mobile Music Studies}, year = {2010}, publisher = {Oxford University Press}, organization = {Oxford University Press}, abstract = {

The Mobile Phone Orchestra (MoPhO) is a new repertoire-based ensemble using mobile phones as the primary musical instrument. While mobile phones have been used for artistic expression before, MoPhO is the first ensemble of its kind and scale, employing more than a dozen players and mobile phones that serve as compositional, research, performance, and educational platforms. MoPhO was founded in Fall 2007 at Stanford University\’s Center for Computer Research in Music and Acoustics (CCRMA) and performed its debut concert in January 2008.\  Since then it has spawned new ensembles at the University of Michigan, as well as in Berlin and Helsinki, and has performed in Genova, Belfast, Helsinki, San Francisco and Berlin. We chronicle the motivation and the process of creating such an ensemble and the repertoire of MoPhO\’s first performances. MoPhO aims to demonstrate that mobile phones orchestras are interesting technological and artistic opportunities for new types of music-making.

}, keywords = {ccrma, mobile phone orchestra, mopho, stanford}, author = {Ge Wang and Georg Essl and Henri Pentinnen} } @inbook {3451, title = {MPEG Audio Compression Basics}, booktitle = {The MPEG Representation of Digital Media}, year = {2012}, publisher = {Springer}, organization = {Springer}, author = {M. Bosi}, editor = {L. Chiariglione} } @inbook {4505, title = {On-the-fly Programming: Using Code as an Expressive Musical Instrument}, booktitle = {A NIME Reader: Fifteen Years of New Interfaces for Musical Expression}, year = {2017}, note = {\ }, publisher = {Springer}, organization = {Springer}, abstract = {\ }, keywords = {aesthetics, ChucK, computer music performance}, isbn = {978-3-319-47214-0}, url = {http://chuck.stanford.edu/}, author = {Ge Wang and Perry R. Cook}, editor = {Alexander Refsum Jensenius and Michael Lyons} } @inbook { chafe:simulating, title = {Simulating Performance on a Bowed Instrument}, booktitle = {Current Directions in Computer Music}, year = {1989}, month = {05/1988}, publisher = {MIT Press}, organization = {MIT Press}, keywords = {bowed instrument, performance, simulating}, url = {https://ccrma.stanford.edu/files/papers/stanm48.pdf}, author = {Chafe, Chris}, editor = {Pierce, John R. and Mathews, Max Vernon} } @conference { berdahlandsmithcontrol06, title = {Active Damping of a Vibrating String}, booktitle = {Proc.~6th International Symposium on Active Noise and Vibration Control}, year = {2006}, month = {Sept.~18{\textendash}20}, address = {Adelaide, Australia}, author = {Edgar Berdahl and Julius O. Smith} } @conference {3234, title = {Approximating Measured Reverberation Using A Hybrid Fixed/Switched Convolution Structure}, booktitle = {Proc. 13th Int. Conf. Digital Audio Effects (DAFx-10)}, year = {2010}, note = {\ }, month = {09/2010}, address = {Graz, Austria}, abstract = {
An efficient reverberator structure is proposed for approximatingmeasured reverberation. A fixed convolution matching the early\ portion of a measured impulse response is crossfaded with a switched\ convolution reverberator drawing its switched convolution section\ from the late-field of the measured impulse response. In this way,\ the early portion of the measured impulse response is precisely reproduced, and the late-field equalization and decay rates efficiently\ approximated. To use segments of the measured impulse response,\ the switched convolution structure is modified to include a normalization filter to account for the decay of the late-field between the\ nominal fixed/switched crossfade time and the time of the selected\ segment. Further, the measured impulse response late-field is extended below its noise floor in anticipation of the normalization.\ This structure provides psychoacoustically accurate synthesis of\ the measured impulse response using less than half a second of\ convolution, irrespective of the length of the measured impulse response. In addition, the structure provides direct control over the\ equalization and late-field frequency dependent decay rate. Emulations of EMT 140 plate reverberator and marble lobby impulse\ responses are presented.
}, keywords = {artificial reverberator}, author = {Lee, Keun Sup and Nicholas Bryan and Jonathan Abel} } @conference { smithfb09, title = {Audio FFT Filter Banks}, booktitle = {Proc. 12th Int. Conf. Digital Audio Effects (DAFx-09), Como, Italy, September 1{\textendash}4}, year = {2009}, author = {Julius O. Smith} } @conference {4382, title = {Bit Bending: an Introduction}, booktitle = {International Conference on Digital Audio Effects (DAFx-13)}, year = {2013}, note = {\ }, month = {09/2013}, address = {Maynooth, Ireland}, abstract = {\ }, author = {Kurt James Werner and Mayank Sanganeria} } @conference {2563, title = {Building Interactive Networked Musical Environments Using q3osc}, booktitle = {AES 35th International Conference: Audio for Games}, year = {2009}, address = {London, UK}, abstract = {

Interactive networked musical gaming environments designed as control systems for external music and sound programming languages can be built using the q3osc Quake III/ioquake3 gaming mod. Bi-directional support for the Open Sound Control (OSC) messaging protocol compiled into the game engine allows for the real-time tracking, sonification, spatialization and third-party control of game entities, clients and environmental parameters. Reactive audio environments ranging from abstract multi-user musical performance spaces to representative acoustic models of physical space can be constructed using either a standard user-centric audio perspective or a potentially more immersive and inclusive space-centric perspective. Issues of space and perspective are discussed as related to the distribution of performance space and sonified environment across both local and wide-area networks.

}, author = {Rob Hamilton} } @conference {2769, title = {CCRMA Papers Presented at the 1995 International Computer Music Conference, Banff, Canada}, booktitle = {International Computer Music Conference}, number = {STAN-M-91}, year = {1995}, month = {09/1995}, address = {Banff, Canada}, keywords = {icmc}, url = {http://ccrma.stanford.edu/files/papers/stanm91.pdf}, author = {Bauman, Marcia L. and David P. Berners and Chafe, Chris and Cook, Perry and Jaffe, David A. and Kunze, Tobias and Lopez-Lezcano, Fernando and Scavone, Gary P. and Julius O. Smith and Stilson, Tim and Taube, Heinrich and Trautmann, Steven D. and Van Duyne, Scott A.} } @conference {2684, title = {CCRMA Papers Presented at the 1996 International Computer Music Conference Hong Kong}, booktitle = {International Computer Music Conference Hong Kong}, number = {STAN-M-99}, year = {1996}, month = {08/1996}, publisher = {CCRMA}, organization = {CCRMA}, address = {Hong Kong}, keywords = {International computer music conference}, url = {https://ccrma.stanford.edu/files/papers/stanm99.pdf}, author = {Chafe, Chris and Igoudin, Alex and Jaffe, David A. and Karjalainen, Matti and Kunze, Tobias and Levine, Scott and Lopez-Lezcano, Fernando and O{\textquoteright}Modhrain, Sile and Porcaro, Nick and Putnam, William and Scandalis, Pat and Scavone, Gary P. and Julius O. Smith and Stilson, Tim and Taube, Heinrich and van Duyne, Scott} } @conference {2685, title = {CCRMA Papers Presented at the 1997 International Computer Music Conference Thessaloniki, Greece}, booktitle = {International Computer Music Conference Thessaloniki, Greece}, number = {STAN-M-101}, year = {1997}, month = {08/1997}, publisher = {CCRMA}, organization = {CCRMA}, address = {Stanford, CA}, keywords = {CCRMA Papers, International}, url = {https://ccrma.stanford.edu/files/papers/stanm101.pdf}, author = {Jonathan Berger and Chafe, Chris and Igoudin, Alex and Jaffe, David A. and Kunze, Tobias and Levine, Scott and Lopez-Lezcano, Fernando and O{\textquoteright}Modhrain, Sile and Scandalis, Pat and Scavone, Gary P. and Julius O. Smith and Stilson, Tim and Taube, Heinrich and van Duyne, Scott and Verma, Tony} } @conference {2821, title = {CCRMA Studio Report}, booktitle = {International Computer Music Conference}, year = {1997}, month = {1997}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {Thessaloniki, Greece}, keywords = {studio report}, url = {https://ccrma.stanford.edu/~nando/publications/ccrma_studio_report_icmc1997.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {2822, title = {CCRMA Studio Report}, booktitle = {International Computer Music Conference}, year = {1998}, month = {1998}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {University of Michigan, Ann Arbor, USA}, keywords = {studio report}, url = {https://ccrma.stanford.edu/~nando/publications/ccrma_studio_report_icmc1998.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {2823, title = {CCRMA Studio Report}, booktitle = {International Computer Music Conference}, year = {1999}, month = {1999}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {Beijing, China}, keywords = {studio report}, url = {https://ccrma.stanford.edu/~nando/publications/ccrma_studio_report_icmc1999.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {2824, title = {CCRMA Studio Report}, booktitle = {International Computer Music Conference}, year = {2001}, month = {2001}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {La Habana, Cuba}, keywords = {studio report}, url = {https://ccrma.stanford.edu/~nando/publications/ccrma_studio_report_icmc2001.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {2825, title = {CCRMA Studio Report}, booktitle = {International Computer Music Conference}, year = {2004}, month = {2004}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {Miami, USA}, keywords = {studio report}, url = {https://ccrma.stanford.edu/~matt/studioreport/CCRMA-studioreport.pdf}, author = {Wright, Matthew and Jonathan Berger and Burns, Chris and Lopez-Lezcano, Fernando and Julius O. Smith} } @conference {2827, title = {CCRMA Studio Report}, booktitle = {International Computer Music Conference}, year = {2007}, month = {2007}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {Copenhagen, Denmark}, keywords = {studio report}, url = {https://ccrma.stanford.edu/~nando/publications/ccrma_studio_report_icmc2007.pdf}, author = {Lopez-Lezcano, Fernando and Wilkerson, Carr} } @conference {2581, title = {CCRMA Studio Report for ICMC 2009}, booktitle = {International Computer Music Conference}, year = {2009}, abstract = {The CCRMA Studio Report attempts to summarize the going\’s on at the Center for Computer Research in Music and Acoustics during 2008-2009 academic year.}, author = {Wilkerson, Carr} } @conference {3473, title = {Clustering and Synchronizing Multi-Camera Video via Landmark Cross-Correlation}, booktitle = {IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP)}, year = {2012}, note = {\ }, month = {03/2012}, publisher = {IEEE}, organization = {IEEE}, address = {Kyoto, Japan}, abstract = {\ }, author = {Nicholas Bryan and Paris Smaragdis and Gautham Mysore} } @conference {2586, title = {Combining Analysis and Synthesis in the ChucK Programming Language}, booktitle = {International Computer Music Conference}, year = {2007}, address = {Copenhagen}, abstract = {

In this paper, we present a new programming model for performing audio analysis, spectral processing, and feature extraction in the ChucK programming language. The solution unifies analysis and synthesis in the same high-level, strongly-timed, and concurrent environment, extending and fully integrating with the existing language framework. In particular, we introduce the notion of a Unit Analyzer (UAna) and new constructs for dataflow, data types and semantics for operations in analysis domains, and mechanisms for seamlessly combining analysis and synthesis tasks in a precisem, sample-synchronous manner. We present the motivation of our system, and describe new language-level syntaxes, semantics, and the underlying implementation. We provide code examples and discuss potential uses and benefits of the system for audio researchers, performers, and teachers.

}, keywords = {analysis, audio, ChucK, synthesis, uana}, url = {http://soundlab.cs.princeton.edu/publications/uana_icmc2007.pdf}, author = {Ge Wang and Rebecca Fiebrink and Cook, Perry} } @conference {3468, title = {A Configurable Microphone Array with Acoustically Transparent Omnidirectional Elements}, booktitle = {127th Audio Engineering Society Convention}, year = {2009}, note = {\ }, month = {09/2009}, publisher = {Audio Engineering Society}, organization = {Audio Engineering Society}, address = {New York City, New York}, abstract = {An acoustically transparent, configurable microphone array with omnidirectional elements, designed for room acoustics analysis and synthesis, and archaeological acoustics applications, is presented. Omnidirectional microphone elements with 2 mm-diameter capsules and 1 mm-diameter wire mounts produce a nearly acoustically transparent array, and provide a simplified mathematical framework for processing measured signals. The wire mounts are fitted onto a 1.6 cm-diameter tube forming the microphone stand, with the microphones arranged above the tube so that acoustic energy can propagate freely across the array. The wire microphone mounts have some flexibility, and the array may be configured. Detachable arms with small speakers are used to estimate the element positions with an accuracy better than the 2 mm microphone diameter.}, author = {Abel, Jonathan S and Nicholas Bryan and Skare, Travis and Miriam Kolar and Patty Huang and Mostowfi, Darius and Julius O. Smith} } @conference {dumitrescu_et_al:DSP:2009:1971, title = {The Dagstuhl Core}, booktitle = {Knowledge representation for intelligent music processing}, series = {Dagstuhl Seminar Proceedings}, number = {09051}, year = {2009}, note = {\ }, publisher = {Schloss Dagstuhl - Leibniz-Zentrum fuer Informatik, Germany}, organization = {Schloss Dagstuhl - Leibniz-Zentrum fuer Informatik, Germany}, address = {Dagstuhl, Germany}, abstract = {\ }, issn = {1862-4405}, url = {http://drops.dagstuhl.de/opus/volltexte/2009/1971}, author = {Dumitrescu, T. and Kepper, J. and Kornstaedt, A. and R{\"o}wenstruck, D. and Roland, P. and Sapp, C. and Selfridge-Field, E.}, editor = {Selfridge-Field, E. and Wiering, F. and Wiggins, G.} } @conference {4123, title = {Design Principles for Lumped Model Discretisation Using M{\"o}bius Transforms}, booktitle = {International Conference on Digital Audio Effects (DAFx-15)}, year = {2015}, note = {\ }, month = {November/2015}, address = {Trondheim, Norway}, abstract = {\ }, url = {http://www.ntnu.edu/web/dafx15/program}, author = {Fran{\c c}ois G. Germain and Kurt James Werner} } @conference {2587, title = {Designing Smule{\textquoteright}s iPhone Ocarina}, booktitle = {New Interfaces for Musical Expression}, year = {2009}, address = {Pittsburgh}, abstract = {

The Smule Ocarina is a wind instrument designed for the\ iPhone, fully leveraging its wide array of technologies:\ microphone input (for breath input), multitouch (for\ fingering), accelerometer, real-time sound synthesis, high-performance graphics, GPS/location, and persistent data\ connection. \ In this mobile musical artifact, the\ interactions of the ancient flute-like instrument are both\ preserved and transformed via breath-control and\ multitouch finger-holes, while the onboard global\ positioning and persistent data connection provide the\ opportunity to create a new social experience, allowing\ the users of Ocarina to listen to one another. \ In this way,\ Ocarina is also a type of social instrument that enables a\ different, perhaps even magical, sense of global\ connectivity.

}, keywords = {ccrma, design, iphone, music, ocarina, smule, social}, url = {http://ccrma.stanford.edu/~ge/publish/ocarina-nime2009.pdf}, author = {Ge Wang} } @conference {4121, title = {Digitizing the Ibanez Weeping Demon Wah Pedal}, booktitle = {International Conference on Digital Audio Effects (DAFx-15)}, year = {2015}, note = {\ }, month = {November/2015}, address = {Trondheim, Norway}, abstract = {\ }, url = {http://www.ntnu.edu/web/dafx15/program}, author = {Chet Gnegy and Kurt James Werner} } @conference {2568, title = {Discrete Time Emulation of the Leslie Speaker}, booktitle = {127th Audio Engineering Society Convention}, year = {2009}, month = {10/2009}, address = {New York, NY, USA}, abstract = {

A discrete-time emulation of the Leslie speaker acoustics is described. The midrange horn and subwoofer baffle are individually modeled, with their rotational dynamics separately tracked, and used to drive time- varying FIR filters applied to the input. The rotational speeds of the horn and baffle are modeled using FIR filters having different shapes for acceleration and deceleration. Several time-varying FIR filter methods were explored, all based on impulse responses tabulated over a dense set of horn and baffle angles. In one method, the input sample scales an interpolated impulse response at the current horn or baffle angle, which is added to the output. An example model of a Leslie 44W is presented.

}, keywords = {Leslie FIR}, author = {Jorge Herrera and Hanson, Craig and Jonathan Abel} } @conference {4122, title = {Distortion and Pitch Processing Using a Modal Reverberator Architecture}, booktitle = {International Conference on Digital Audio Effects (DAFx-15)}, year = {2015}, note = {\ }, month = {November/2015}, address = {Trondheim, Norway}, abstract = {\ }, url = {http://www.ntnu.edu/web/dafx15/program}, author = {Abel, Jonathan S and Kurt James Werner} } @conference {2679, title = {DMIX- A Multi Faceted Environment for Composing and Performing Computer Music: Its Design, Philosophy, and Implementation}, booktitle = {Fourth Biennial Arts and Technology Symposium}, year = {1993}, month = {10/1993}, publisher = {CCRMA}, organization = {CCRMA}, address = {Stanford, CA}, keywords = {composing, computer music, design, dmix, Environment, Performing, Philosophy}, issn = {STAN-M-83}, url = {https://ccrma.stanford.edu/files/papers/stanm83.pdf}, author = {Oppenheim, Daniel V.} } @conference {2574, title = {A dynamic spatial locator ugen for CLM}, booktitle = {Sound and Music Computing Conference}, year = {2008}, month = {07/2008}, address = {Berlin, Germany}, abstract = {Dlocsig is a dynamic spatial locator unit generator written for the Common Lisp Music (CLM) sound
synthesis and processing language. Dlocsig was first created in 1992 as a four channel 2d dynamic locator and since then it has evolved to a full 3d system for an arbitrary number of speakers that can render moving sound objects through amplitude panning (VBAP) or Ambisonics. This paper describes the motivations for the project, its evolution over time and the details of its software implementation and user interface.}, url = {https://ccrma.stanford.edu/~nando/publications/dlocsig_smc2008.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {3614, title = {An Efficient Posterior Regularized Latent Variable Model for Interactive Sound Source Separation}, booktitle = {International Conference on Machine Learning}, year = {2013}, note = {\ }, month = {06/2013}, address = {Atlanta, GA}, abstract = {\ }, author = {Nicholas Bryan and Gautham Mysore} } @conference { cassidyandsmith07, title = {Efficient Time-Varying Loudness Estimation via the Hopping Goertzel DFT}, booktitle = {Proceedings of the IEEE International Midwest Symposium on Circuits and Systems (MWSCAS-2007)}, year = {2007}, author = {Ryan J. Cassidy and Julius O. Smith} } @conference {2643, title = {An electroacoustic sound transmission system that is stable in any (dissipative) acoustic environment: An application of sound portholes}, booktitle = {NOISE-CON 2010 in conjunction ith the 159th Meeting of the Acoustical Society of America}, year = {2010}, month = {04/2010}, address = {Baltimore, MD}, abstract = {Active electroacoustic systems are commonly employed to transmit sound from one
location to another. For example, consider the following configuration: a person talks into
a microphone, which produces an electrically amplified signal for driving a loudspeaker,
and the loudspeaker induces an output acoustic signal allowing someone else at a more
distant location to hear the person talking. Unfortunately, such system designs typically do
not consider acoustic feedback, which can destabilize the system and result in \“howling.\”
In contrast, a feedback control system can transmit sound from one location to another
without the risk of howling using sound portholes, which are collocated
microphone/loudspeaker transducers. We design feedback controllers to transmit sound
between sound portholes.\  These controllers model physical analog systems, such as a
spring or a gyrator. The spring controller essentially binds the diaphragms of the two
sound portholes together. We relate the spring controller to the string connecting two tin
cans in the classical tin can telephone. Measurements are performed on a real feedback
control system with two sound portholes. Because the feedback controller models a passive
system, it is theoretically stable in any (dissipative) acoustic environment.
}, url = {https://ccrma.stanford.edu/~eberdahl/Papers/NOISECON2010Communication.pdf}, author = {Edgar Berdahl and Dan Harris and Gunter Niemeyer and Julius O. Smith} } @conference {4115, title = {An Energetic Interpretation of Nonlinear Wave Digital Filter Lookup Table Error}, booktitle = {IEEE International Symposium on Signals, Circuits, and Systems (ISSCS)}, year = {2015}, note = {\ }, month = {July/2015}, address = {Iasi, Romania}, abstract = {\ }, author = {Kurt James Werner and Smith III, Julius O.} } @conference {4705, title = {Equalization matching of speech recordings in real-world environments}, booktitle = {IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP)}, year = {2016}, note = {\ }, month = {03/2016}, publisher = {IEEE}, organization = {IEEE}, address = {Shanghai, China}, abstract = {When different parts of speech content such as voice-overs and narration are recorded in real-world environments with different acoustic properties and background noise, the difference in sound quality between the recordings is typically quite audible and therefore undesirable. We propose an algorithm to equalize multiple such speech recordings so that they sound like they were recorded in the same environment. As the timbral content of the speech and background noise typically differ considerably, a simple equalization matching results in a noticeable mismatch in the output signals. A single equalization filter affects both timbres equally and thus cannot disambiguate the competing matching equations of each source. We propose leveraging speech enhancement methods in order to separate speech and background noise, independently apply equalization filtering to each source, and recombine the outputs. By independently equalizing the separated sources, our method is able to better disambiguate the matching equations associated with each source. Therefore the resulting matched signals are perceptually very similar. Additionally, by retaining the background noise in the final output signals, most artifacts from speech enhancement methods are considerably reduced and in general perceptually masked. Subjective listening tests show that our approach significantly outperforms simple equalization matching.
Audio examples}, isbn = {978-1-4799-9988-0}, doi = {10.1109/ICASSP.2016.7471747}, url = {http://ieeexplore.ieee.org/document/7471747/}, author = {Fran{\c c}ois G. Germain and Mysore, Gautham J. and Takako Fujioka} } @conference {3474, title = {Evolving The Mobile Phone Orchestra}, booktitle = {International Conference on New Interfaces for Musical Expression}, year = {2010}, note = {\ }, month = {06/2010}, address = {Sydney, Australia}, abstract = {\ }, author = {Jieun Oh and Jorge Herrera and Nicholas Bryan and Ge Wang} } @conference {4338, title = {The Fender Bassman 5F6-A Family of Preamplifier Circuits{\textemdash}A Wave Digital Filter Case Study}, booktitle = {International Conference on Digital Audio Effects (DAFx-16)}, year = {2016}, note = {\ }, month = {09/2016}, address = {Brno, Czech Republic}, abstract = {\ }, url = {http://dafx16.vutbr.cz/}, author = {W. Ross Dunkel and Maximilian Rest and Kurt James Werner and Michael J. Olsen and Smith III, Julius O.} } @conference {4703, title = {Fixed-rate Modeling of Audio Lumped Systems: A Comparison Between Trapezoidal and Implicit Midpoint Methods}, booktitle = {20th International Conference on Digital Audio Effects (DAFx-17)}, year = {2017}, note = {\ }, month = {09/2017}, address = {Edinburgh, UK}, abstract = {This paper presents the implicit midpoint method as an alternative to the more typical trapezoidal method for the simulation of audio lumped systems at a fixed rate. We provide tools for understanding the behavior and error associated with each method in the case of a typical nonlinear state-space formulation. We also show implementation strategies for those methods, including how an implicit midpoint method solution can be generated from a trapezoidal method solution and vice versa. Finally, we present some empirical analysis of the behavior of each method for a simple diode clipper circuit and provide an approach to help interpret their relative performance and how to pick the more appropriate method depending on desirable properties. The presented tools are also intended as a general approach to interpret the performance of discretization approaches.}, url = {http://www.dafx17.eca.ed.ac.uk/papers/DAFx17_paper_90.pdf}, author = {Fran{\c c}ois G. Germain} } @conference {3449, title = {From Jack to UDP packets to sound, and back}, booktitle = {Linux Audio Conference 2012}, year = {2012}, month = {04/2012}, publisher = {CCRMA, Stanford University}, organization = {CCRMA, Stanford University}, address = {CCRMA, Stanford University}, url = {https://ccrma.stanford.edu/~nando/publications/jack_mamba_lac2012.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {4118, title = {A General and Explicit Formulation for Wave Digital Filters with Multiple/Multiport Nonlinearities and Complicated Topologies}, booktitle = {IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA)}, year = {2015}, note = {\ }, month = {October/2015}, address = {New Paltz, New York}, abstract = {\ }, author = {Kurt James Werner and Vaibhav Nangia and Smith III, Julius O. and Abel, Jonathan S} } @conference {4491, title = {Generalizing Root Variable Choice in Wave Digital Filters with Grouped Nonlinearities}, booktitle = {20th International Conference on Digital Audio Effects (DAFx-17)}, year = {2017}, note = {\ }, month = {09/2017}, address = {Edinburgh, UK}, abstract = {\ }, author = {Kurt James Werner and Olsen, Michael J{\o}rgen and Maximilian Rest and Parker, Julian D.} } @conference {4383, title = {Grain Proc: a Real-Time Granular Synthesis Interface for Live Performance}, booktitle = {NIME}, year = {2013}, note = {\ }, month = {05/2013}, address = {Daejeon, Korea}, abstract = {\ }, author = {Mayank Sanganeria and Kurt James Werner} } @conference {2558, title = {The GRIP MAESTRO: Balancing Large- and Small-scale Aspects of Emotive Gestural Control for Live Electroacoustic Musical Performance}, booktitle = {Hawaii International Conference on the Arts and Humanities}, year = {2010}, address = {Honolulu, Hawaii, USA}, abstract = {\ \ \  The goal of this research is to explore the interrelationships that exist between performative human gestures and musical sound by developing effective and idiomatic software mappings for the GRIP MAESTRO electroacoustic music performance interface.

\ \ \  In the spring of 2008 I created the GRIP MAESTRO as part of my ongoing research in physical interactive design at Stanford University\’s Center for Computer Research in Music and Acoustics (C.C.R.M.A.).\  Unlike many such interfaces that are designed with a specific sound or sonic metaphor in mind, the GRIP MAESTRO was built to take advantage of a specific physical gesture for its emotive potential, with little forethought given to the sounds it would eventually produce.\  The specific gesture of which I am speaking is that of gripping, of squeezing with the entire hand or with individual fingers.\  This gesture is one that is frequently underutilized in musical performance.\  The hand may be used to depress keys or strings, but never is the full range of motion taken advantage of as it is with the GRIP MAESTRO.\  The interface couples this familiar yet nuanced gesture with built-in physical resistance to allow the performer to feel his performance in the same way he would on an acoustic instrument.
\ \ \  Since the GRIP MAESTRO is modeled after a gesture and not after a desired sound (or existing method of sound production), finding/developing a musical sound that works idiomatically for the instrument is the fundamental challenge it poses.\  An ideal software mapping will produce sounds that, to an observer and to the performer, seem to come naturally from the device and the gestures its performance requires.\  This is not to say that there may be only one perfect mapping for the GRIP MAESTRO, on the contrary, the multidimensionality of the challenge ensures the possibility of many such ideals.\  It is therefore this wealth of options that is simultaneously the instrument\’s best strength and its biggest potential hindrance.

\ \ \  In its current incarnation the GRIP MAESTRO consists of two modified hand exercisers, held one in each hand, connected via a microcontroller board to a computer where the information sent by the device\’s sensors is interpreted and converted into sound.\  Each GRIP MAESTRO hand-unit has four independent spring-loaded finger pads and a single three-spring palm pad.\  Attached near each spring is a small yet powerful magnet opposite a Hall Effect sensor.\  The data that each sensor sends out is directly related to its distance from its corresponding magnet, and thereby indicates the amount that each pad is depressed.\  The palm pad has two sensor/magnet pairs so that its \‘tilt\’ maybe calculated by the computer; that is the angle of the palm rest relative to the rest of the unit.\  New to this most recent version of the instrument, each unit is now also equipped with an accelerometer which detects its orientation to gravity and relative motion in three-dimensional space.\  When one compares the sensor readings from one unit against those of the other, one is able to extrapolate even larger pieces of performance information.\ 
\ \ \  The range of gestural motion that the GRIP MAESTRO is able to detect is extremely wide; from the subtlest twitch of a single finger to the wildest most dramatic swing of the arms.\  The challenge of mapping this gesturally-derived information naturally onto meaningful musical sounds becomes one of balancing not only the large- and small-scale aspects of the gestures themselves, but also the gestural information with those similarly varied aspects of musical sound (i.e. large-scale musical aspects such as phrase and melody, and small-scale aspects such as pitch, vibrato, and timbre).

\ \ \  Since the accelerometer-based motion-sensing and two-handed aspects of the GRIP MAESTRO are relatively recent additions to the design, few of my software mappings thus far have utilized them to their fullest potentials.\  It is in this area of the project that the most fruitful and exciting work is yet to be done.\  Already the GRIP MAESTRO has generated creative interest for members of the Stanford Laptop Orchestra, as well as for researchers in methods of physical recovery for stroke victims.\ 
\ \ \  For my poster session at the Hawaii International Conference on Arts and Humanities I will present the work that I have completed so far together with that which has yet to be done.\  In the coming months I will develop several more software mappings for the device and compose a set of etudes through which they may be showcased.\  The GRIP MAESTRO is literally a hands-on experience and interactivity is at the core of its being, it is for these reasons that a poster session is the ideal setting in which to demonstrate the device.}, keywords = {Accellerometer, ChucK, GRIP MAESTRO, Human Computer Interface, Physical Interactive Design}, author = {Michael Berger} } @conference {4119, title = {An Improved and Generalized Diode Clipper Model for Wave Digital Filters}, booktitle = {139th Convention of the Audio Engineering Society (AES)}, year = {2015}, note = {\ }, month = {October/2015}, address = {New York, New York}, abstract = {\ }, author = {Kurt James Werner and Vaibhav Nangia and Alberto Bernardini and Smith III, Julius O. and Augusto Sarti} } @conference { berdahlandsmithifm, title = {Impulse Response Measurement Toolbox}, year = {2008}, note = {http://cnx.org/content/col10519/latest/}, author = {Edgar Berdahl and Julius O. Smith} } @conference {3470, title = {Impulse Response Measurements in the Presence of Clock Drift}, booktitle = {129th Audio Engineering Society Convention}, year = {2010}, note = {\ }, month = {11/2010}, publisher = {Audio Engineering Society}, organization = {Audio Engineering Society}, address = {San Francisco, California}, abstract = {\ }, keywords = {Room Impulse Responses}, author = {Nicholas Bryan and Miriam Kolar and Abel, Jonathan S} } @conference { berdahlandsmith07, title = {Inducing Unusual Dynamics in Acoustic Musical Instruments}, booktitle = {Proc. IEEE International Conference on Control Applications, Singapore, Oct. 1{\textendash}3}, year = {2007}, pages = {1336{\textendash}1341}, author = {Edgar Berdahl and Julius O. Smith} } @conference { arvindhandsmith03, title = {Inferring Control Inputs to an Acoustic Violin from Audio Spectra}, booktitle = {Proc. Int. Conf. Multimedia Eng.}, year = {2003}, publisher = {IEEE Press}, organization = {IEEE Press}, address = {New York}, author = {Arvindh Krishnaswamy and Julius O. Smith} } @conference {3613, title = {Interactive Refinement of Supervised and Semi-Supervised Sound Source Separation Estimates}, booktitle = {IEEE International Conference on Acoustics, Speech, and Signal Processing}, year = {2013}, note = {\ }, month = {05/2013}, address = {Vancouver, Canada}, abstract = {\ }, keywords = {source separation}, author = {Nicholas Bryan and Gautham Mysore} } @conference {3612, title = {Interactive User-Feedback for Sound Source Separation}, booktitle = {International Conference on Intelligent User-Interfaces (IUI), Workshop on Interactive Machine Learning}, year = {2013}, note = {\ }, month = {03/2013}, address = {Santa Monica}, abstract = {\ }, author = {Nicholas Bryan and Gautham Mysore} } @conference {3753, title = {ISSE: An Interactive Source Separation Editor}, booktitle = {CHI Conference on Human Factors in Computing Systems}, year = {2014}, note = {\ }, month = {04/2014}, publisher = {ACM}, organization = {ACM}, address = {Toronto, Canada}, abstract = {\ }, author = {Nicholas Bryan and Mysore, Gautham J. and Ge Wang} } @conference { caceres2009:jacktripsoundwire, title = {JackTrip/SoundWIRE Meets Server Farm}, booktitle = {Proceedings of 6th Sound and Music Computing Conference}, year = {2009}, pages = {95{\textendash}98}, address = {Porto, Portugal}, author = {C{\'a}ceres, Juan-Pablo and Chafe, Chris} } @conference {4489, title = {Joint Parameter Optimization of Differentiated Discretization Schemes for Audio Circuits}, booktitle = {142nd International Convention of the Audio Engineering Society}, year = {2017}, note = {\ }, month = {05/2017}, address = {Berlin, Germany}, abstract = {\ }, author = {Fran{\c c}ois G. Germain and Kurt James Werner} } @conference {3480, title = {Learnng Sparse Feature Representations For Music Annotation And Retrieval}, booktitle = {13th International Society for Music Information Retrieval Conference}, year = {2012}, note = {\ }, month = {10/2012}, address = {Porto, Portugal}, abstract = {We present a data-processing pipeline based on sparse feature learning and describe its applications to music annotation and retrieval. Content-based music annotation and retrieval systems process audio starting with features. While commonly used features, such as MFCC, are handcrafted to extract characteristics of the audio in a succinct way, there is increasing interest in learning features automatically from data using unsupervised algorithms. We describe a systemic approach applying feature-learning algorithms to music data, in particular, focusing on a highdimensional sparse-feature representation. Our experiments show that, using only a linear classifier, the newly learned features produce results on the CAL500 dataset comparable to state-of-the-art music annotation and retrieval systems.}, url = {http://ccrma.stanford.edu/~juhan/pubs/jnam-ismir2012.pdf}, author = {Juhan Nam and Jorge Herrera and Slaney, Malcolm and Smith, Julius} } @conference { huangandsmith04, title = {Loudness-Based Display and Analysis Applied to Artificial Reverberation}, booktitle = {Proc. 2004 Int. Computer Music Conf., Miami, Florida}, year = {2004}, month = {nov}, publisher = {Computer Music Association}, organization = {Computer Music Association}, author = {Patty Huang and Julius O. Smith} } @conference {2645, title = {A Method for Estimating Interaural Time Difference for Binaural Synthesis}, booktitle = {the 125th Audio Engineering Society Convention}, year = {2008}, note = {\ }, month = {10/2008}, publisher = {AES}, organization = {AES}, address = {San Francisco, CA, USA}, abstract = {\ A method for estimating interaural time difference (ITD) from measured head-related transfer functions (HRTFs) is presented. The method forms ITD as the difference in left-ear and right-ear arrival times, estimated as the times of maximum cross-correlation between measured HRTFs and their minimum-phase counterparts. This arrival time estimate is related to a nonlinear least-squares fit to the measured excess phase, emphasizing those frequencies having large HRTF magnitude and deweighting large phase delay errors. As HRTFs are nearly minimum-phase, this method is robust compared to the conventional approach of cross-correlating left-ear and right-ear HRTFs, which can be very different. The method also performs slightly better than techniques averaging phase delay over a limited frequency range.}, keywords = {acoustic signal processing, HRTF, ITD}, url = {http://www.aes.org/e-lib/browse.cfm?elib=14763}, author = {Juhan Nam and Jonathan Abel and Julius O. Smith} } @conference {3471, title = {Methods For Extending Room Impulse Responses Beyond Their Noise Floor}, booktitle = {129th Audio Engineering Society Convention}, year = {2010}, note = {\ }, month = {11/2010}, publisher = {Audio Engineering Society}, organization = {Audio Engineering Society}, address = {San Francisco, California}, abstract = {\ }, author = {Nicholas Bryan and Abel, Jonathan S} } @conference { arvindhandsmithmohonk03, title = {Methods for Simulating String Collisions with Rigid Spatial Objects}, booktitle = {Proc. IEEE Workshop on Appl. Signal Processing to Audio and Acoustics, New Paltz, NY}, year = {2003}, month = {oct}, publisher = {IEEE Press}, organization = {IEEE Press}, address = {New York}, author = {Arvindh Krishnaswamy and Julius O. Smith} } @conference {4488, title = {Modal Analysis and Synthesis of the University of Michigan Lurie Carillon}, booktitle = {Resonance and Remembrance: An Interdisciplinary Bell Studies Symposium}, year = {2017}, note = {\ }, month = {03/2017}, address = {University of Michigan, Ann Arbor, MI}, abstract = {\ }, url = {https://gobluebells.wordpress.com/2016/09/21/cfp-resonance-remembrance/}, author = {Kermit-Canfield, Elliot and Kurt James Werner} } @conference {4490, title = {Modal Audio Effect: A Carillon Case Study}, booktitle = {20th International Conference on Digital Audio Effects (DAFx-17)}, year = {2017}, note = {\ }, month = {09/2017}, address = {Edinburgh, UK}, abstract = {\ }, author = {Kermit-Canfield, Elliot and Kurt James Werner} } @conference {4116, title = {Modeling a Class of Multi-Port NonLinearities in Wave Digital Structures}, booktitle = {European Signal Processing Conference (EUSIPCO)}, year = {2015}, note = {\ }, month = {August/2015}, address = {Nice, France}, abstract = {\ }, author = {Alberto Bernardini and Kurt James Werner and Augusto Sarti and Smith III, Julius O.} } @conference { huangserafinandsmithdafx00, title = {Modeling High-Frequency Modes of Complex Resonators Using a Waveguide Mesh}, booktitle = {Proc. Int. Conf. Digital Audio Effects (DAFx-00), Verona, Italy}, year = {2000}, note = {http://www.dafx.de/}, author = {Patty Huang and Stefania Serafin and Julius O. Smith} } @conference {4340, title = {Modeling the Hammond Organ Vibrato/Chorus with Wave Digital Filters}, booktitle = {International Conference on Digital Audio Effects (DAFx-16)}, year = {2016}, note = {\ }, month = {09/2016}, address = {Brno, Czech Republic}, abstract = {\ }, url = {http://dafx16.vutbr.cz/}, author = {Kurt James Werner and W. Ross Dunkel and Fran{\c c}ois G. Germain} } @conference {3475, title = {MoMu: A Mobile Music Toolkit}, year = {2010}, note = {\ }, month = {06/2010}, address = {Sydney, Australia}, abstract = {\ }, author = {Nicholas Bryan and Jorge Herrera and Jieun Oh and Ge Wang} } @conference {3834, title = {More Cowbell: a Physically-Informed, Circuit-Bendable, Digital Model of the TR-808 Cowbell}, booktitle = {137th International Audio Engineering Society (AES) Conference}, year = {2014}, note = {\ }, month = {10/2014}, address = {Los Angeles, CA}, abstract = {\ }, keywords = {cowbell, drum machine, physical modeling, TR-808, virtual analog}, url = {http://www.aes.org/e-lib/browse.cfm?elib=17530}, author = {Kurt James Werner and Abel, Jonathan S and Julius O. Smith} } @conference {4114, title = {Multi-Port NonLinearities in Wave Digital Structures}, booktitle = {IEEE International Symposium on Signals, Circuits, and Systems (ISSCS)}, year = {2015}, note = {\ }, month = {July/2015}, address = {Iasi, Romania}, abstract = {\ }, author = {Alberto Bernardini and Kurt James Werner and Augusto Sarti and Smith III, Julius O.} } @conference {3477, title = {Musical Influence Network Analysis and Rank of Sampled-Based Music}, booktitle = {12th International Society for Music Information Retrieval Conference}, year = {2011}, note = {\ }, month = {09/2011}, address = {Miami, Florida}, abstract = {\ }, author = {Nicholas Bryan and Ge Wang} } @conference {2755, title = {Noise and Aperiodicity in the Glottal Source: A Study of Singer Voices}, booktitle = {Twelfth International Congress of Phoenetic Sciences}, number = {STAN-M-75}, year = {1991}, month = {08/1991}, keywords = {aperiodicity, glottal source, noise, voices}, url = {https://ccrma.stanford.edu/files/papers/stanm75.pdf}, author = {Cook, Perry} } @conference {3197, title = {Non-negative Hidden Markov Modeling of Audio with Application to Source Separation}, booktitle = {International Conference on Latent Variable Analysis and Signal Separation (LVA / ICA)}, year = {2010}, month = {09/2010}, address = {St. Malo, France}, abstract = {In recent years, there has been a great deal of work in modeling audio using non-negative matrix factorization and its probabilistic counterparts as they yield rich models that are very useful for source separation and automatic music transcription. Given a sound source, these algorithms learn a dictionary of spectral vectors to best explain it. This dictionary is however learned in a manner that disregards a very important aspect of sound, its temporal structure. We propose a novel algorithm, the non-negative hidden Markov model (N-HMM), that extends the aforementioned models by jointly learning several small spectral dictionaries as well as a Markov chain that describes the structure of changes between these dictionaries. We also extend this algorithm to the non-negative factorial hidden Markov model (N-FHMM) to model sound mixtures, and demonstrate that it yields superior performance in single channel source separation tasks.
}, url = {https://ccrma.stanford.edu/~gautham/Site/NFHMM.html}, author = {Mysore, Gautham J. and Paris Smaragdis and Bhiksha Raj} } @conference {3469, title = {On Estimating Room Impulse Responses From Recorded Balloon Pops}, booktitle = {129th Audio Engineering Society Convention}, year = {2010}, note = {\ }, month = {11/2010}, publisher = {Audio Engineering Society }, organization = {Audio Engineering Society }, address = {San Francisco, California}, abstract = {Balloon pops are convenient for probing the acoustics of a space, as they generate relatively uniform radiation patterns and consistent \"N-wave\" waveforms. However, the N-wave spectrum contains nulls which impart an undesired comb-filter-like quality when the recorded balloon pop is convolved with audio. Here, a method for converting recorded balloon pops into full audio bandwidth impulse responses is presented. Rather than directly processing the balloon pop recording, an impulse response is synthesized according to the echo density and frequency band energies estimated in running windows over the balloon pop. Informal listening tests show good perceptual agreement between measured room impulse responses using a loudspeaker source and a swept sine technique, and those derived from recorded balloon pops.}, keywords = {Balloon Pops, N-Wave, Room Impulse Responses}, author = {Abel, Jonathan S and Nicholas Bryan and Patty Huang and Miriam Kolar and Bissera Pentcheva} } @conference {2644, title = {On the Minimum-phase Nature of Head-Related Transfer Functions}, booktitle = {the 125th Audio Engineering Society Convention}, year = {2008}, note = {\ }, month = {10/2008}, publisher = {AES}, organization = {AES}, address = {San Francisco, CA, USA}, abstract = {\ For binaural synthesis, head-related transfer functions (HRTFs) are commonly implemented as pure delays followed by minimum-phase systems. Here, the minimum-phase nature of HRTFs is studied. The cross-coherence between minimum-phase and unprocessed measured HRTFs was seen to be greater than 0.9 for a vast majority of the HRTFs, and was rarely below 0.8. Non-minimum-phase filter components resulting in reduced cross-coherence appeared in frontal and ipsilateral directions. The excess group delay indicates that these non-minimum-phase components are associated with regions of moderate HRTF energy. Other regions of excess phase correspond to high-frequency spectral nulls, and have little effect on cross-coherence.}, keywords = {binaural sound, HRTF, minimum-phase system}, url = {http://www.aes.org/e-lib/browse.cfm?elib=14698}, author = {Juhan Nam and Miriam Kolar and Jonathan Abel} } @conference {2653, title = {Openmixer: a routing mixer for multichannel studios}, booktitle = {Linux Audio Conference 2010}, year = {2010}, month = {5/2010}, address = {Utrech, The Netherlands}, abstract = {The Listening Room at CCRMA, Stanford University is a 3D studio with 16 speakers (4 hang
from the ceiling, 8 surround the listening area at ear level and 4 more are below an acoustically
transparent grid floor). We found that a standard commercial digital mixer was not the best interface
for using the studio. Digital mixers are complex, have an opaque interface and they are usually
geared towards mixdown to stereo instead of efficiently routing many input and output channels.
We have replaced the mixer with a dedicated computer running Openmixer, an open source
custom program designed to mix and route many input channels into the multichannel speaker array
available in the Listening Room. This paper will describe Openmixer, its motivations, current status
and future planned development.
}, url = {https://ccrma.stanford.edu/~nando/publications/openmixer_lac2010.pdf}, author = {Lopez-Lezcano, Fernando and Sadural, Jason} } @conference {4492, title = {Optimizing Differentiated Discretization for Audio Circuits Beyond Driving Point Transfer Functions}, booktitle = {IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA)}, year = {2017}, note = {\ }, month = {10/2017}, address = {New Paltz, New York}, abstract = {\ }, author = {Fran{\c c}ois G. Germain and Kurt James Werner} } @conference { cassidyandsmith04, title = {An oversampled, non-uniform filter bank for multi-band audition and level modification of audio signals}, booktitle = {Proc. 38th Asilomar Conference on Signals, Systems and Computers}, year = {2004}, author = {Ryan J. Cassidy and Julius O. Smith} } @conference {2819, title = {PadMaster: an improvisation environment for real time performance}, booktitle = {International Computer Music Conference}, year = {1995}, month = {1995}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {Banff Centre for the Arts, Canada}, keywords = {improvisation, radio drum, real-time, software}, url = {https://ccrma.stanford.edu/~nando/publications/padmaster_icmc1995.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {2820, title = {PadMaster: banging on algorithms with alternative controllers}, booktitle = {International Computer Music Conference}, year = {1996}, month = {1996}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {Hong Kong University of Science and Technology, China}, keywords = {improvisation, radio drum, real-time, software-design}, url = {https://ccrma.stanford.edu/~nando/publications/padmaster_icmc1996.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {3832, title = {A Physically-Informed, Circuit-Bendable, Digital Model of the Roland TR-808 Bass Drum Circuit}, booktitle = {17th International Conference on Digital Audio Effects (DAFx-14)}, year = {2014}, note = {\ }, month = {09/2014}, address = {Erlangen, Germany}, abstract = {\ }, keywords = {bass drum, drum machine, physical modeling, TR-808, virtual analog}, url = {http://www.dafx14.fau.de/papers/dafx14_kurt_james_werner_a_physically_informed,_ci.pdf}, author = {Kurt James Werner and Abel, Jonathan S and Julius O. Smith} } @conference {2826, title = {The Planet CCRMA software collection}, booktitle = {International Computer Music Conference}, year = {2002}, month = {2002}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {G{\"o}teborg, Sweden}, abstract = {In this paper I would like to present an overview of Planet CCRMA, a freely available collection of Open Source sound and music packages that reflect the linux software environment I currently maintain at CCRMA. It is build on top of RedHat 7.2 and is easy to install and upgrade. Land on Planet CCRMA at http://ccrma.stanford.edu/planetccrma/software/}, keywords = {open source, package collection}, url = {https://ccrma.stanford.edu/~nando/publications/planetccrma_icmc2002.pdf}, author = {Lopez-Lezcano, Fernando} } @conference { chafeicassp90, title = {Pulsed Noise in Self-Sustained Oscillations of Musical Instruments}, booktitle = {Proceddings of the International Conference on Acoustics, Speech, and Signal Processing, Albuquerque}, number = {STAN-M-61}, year = {1990}, month = {01/1990}, publisher = {IEEE Press}, organization = {IEEE Press}, address = {New York}, keywords = {musical instruments, oscillations, pulsed noise, self-sustaining}, url = {https://ccrma.stanford.edu/files/papers/stanm61.pdf}, author = {Chafe, Chris} } @conference {2576, title = {The Quest for Noiseless Computers}, booktitle = {Linux Audio Conference}, year = {2009}, month = {04/2009}, address = {Parma, Italy}, abstract = {When The Knoll, the building that houses CCRMA, was completely renovated in 2004, new custom workstations were designed and built with the goal of being both fast machines and completely noiseless to match the architectural and acoustical design of the building.}, url = {https://ccrma.stanford.edu/~nando/publications/noiseless_computers_lac2009.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {2578, title = {Relative Pitch Estimation of Multiple Instruments}, booktitle = {IEEE International Conference on Acoustics, Speech, and Signal Processing (ICASSP)}, year = {2009}, abstract = {

We present an algorithm based on probabilistic latent component analysis and employ it for relative pitch estimation of multiple instruments in polyphonic music. A multilayered positive deconvolution is performed concurrently on mixture constant-Q transforms to obtain a relative pitch track and timbral signature for each instrument. Initial experimental results on mixtures of two instruments are quite promising and show high levels of accuracy.

}, url = {https://ccrma.stanford.edu/~gautham/Site/Multipitch.html}, author = {Mysore, Gautham J. and Paris Smaragdis} } @conference {4339, title = {Resolving Grouped Nonlinearities in Wave Digital Filters using Iterative Techniques}, booktitle = {International Conference on Digital Audio Effects (DAFx-16)}, year = {2016}, note = {\ }, month = {09/2016}, address = {Brno, Czech Republic}, abstract = {\ }, url = {http://dafx16.vutbr.cz/}, author = {Michael J. Olsen and Kurt James Werner and Smith III, Julius O.} } @conference {4124, title = {Resolving Wave Digital Filters with Multiple/Multiport Nonlinearities}, booktitle = {International Conference on Digital Audio Effects (DAFx-15)}, year = {2015}, note = {\ }, month = {November/2015}, address = {Trondheim, Norway}, abstract = {\ }, url = {http://www.ntnu.edu/web/dafx15/program}, author = {Kurt James Werner and Vaibhav Nangia and Smith III, Julius O. and Abel, Jonathan S} } @conference { abelandsmithclipped, title = {Restoring a Clipped Signal}, booktitle = {Proc. Int. Conf. Acoustics, Speech, and Signal Processing, Toronto}, year = {1991}, publisher = {IEEE Press}, organization = {IEEE Press}, address = {New York}, author = {Jonathan Abel and Julius O. Smith} } @conference {3233, title = {A Reverberator with Two-Stage Decay and Onset Time Controls}, booktitle = {129th Audio Engineering Society Convention}, year = {2010}, note = {\ }, month = {11/2010}, address = {San Francisco, CA, USA}, abstract = {\ An efficient artificial reverberator having two-stage decay and onset time controls is presented. A second-order comb filter controlling the reverberator frequency-dependent decay rates and onset times drives a switched convolution with short noise sequences. In this way, a non-exponential reverberation envelope is produced by the comb filter, while the switched convolution structure produces a high echo density. Several schemes for generating two-stage decays and onset time controls with different onset characteristics in different frequency bands are described.}, keywords = {artificial reverberator}, author = {Lee, Keun Sup and Jonathan Abel} } @conference {4337, title = {RTWDF{\textemdash}A modular wave digital filter library with support for arbitrary topologies and multiple nonlinearities}, booktitle = {International Conference on Digital Audio Effects (DAFx-16)}, year = {2016}, note = {\ }, month = {09/2016}, address = {Brno, Czech Republic}, abstract = {\ }, url = {http://dafx16.vutbr.cz/}, author = {Maximilian Rest and W. Ross Dunkel and Kurt James Werner and Smith III, Julius O.} } @conference {3536, title = {Simulating an Asymmetrically Saturated Nonlinearity Using an LNLNL Cascade}, booktitle = {133rd Audio Engineering Society Convention}, year = {2012}, note = {\ }, month = {10/2012}, address = {San Francisco, CA, USA}, abstract = {The modeling of a weakly nonlinear system having an asymmetric saturating nonlinearity is considered, and a computationally efficient model is proposed. The nonlinear model is the cascade of linear filters and memoryless nonlinearities, an LNLNL system. The two nonlinearities are upward and downward saturators, limiting, respectively, the amplitude of their input for either positive or negative excursions. In this way, distortion noted in each half an input sinusoid can be separately controlled. This simple model is applied toy simulating the signal chain of the Echoplex EP-4 tape delay, where informal listening tests showed excellent agreement between recorded and simulated program material.}, author = {Lee, Keun Sup and Abel, Jonathan S} } @conference { cassidyandsmithmohonk07, title = {Simulation of Recruiting Hearing Impairment Using a Tree-Structured Allpass-Complementary ERB-Band Filter Bank}, booktitle = {IEEE Workshop on Appl. Signal Processing to Audio and Acoustics, Demo Session, New Paltz, NY}, year = {2007}, month = {oct}, pages = {[TD1-8]}, author = {Ryan J. Cassidy and Julius O. Smith} } @conference { gordonandsmith, title = {A Sine Generation Algorithm for VLSI Applications}, booktitle = {Proc. 1985 Int. Computer Music Conf., Vancouver}, year = {1985}, publisher = {Computer Music Association}, organization = {Computer Music Association}, author = {John W. Gordon and Julius O. Smith} } @conference {2557, title = {Skeleton: the Decayed Remnants of Time. An Investigation of Memory Through Sound.}, booktitle = {Hawaii International Conference on the Arts and Humanities}, year = {2009}, address = {Honolulu, Hawaii, USA}, abstract = {\ \ \  The goals of this research are to explore the implications of a new definition of music exemplified in a new composition, and, through its forthcoming premieres, to examine the performances and reception of the work in order to assess the validity of the definition.

\ \ \  In the experimental environment of the early Twentieth Century, the composer Edgard Var\èse described music simply as \“organized sound.\”\  At the time, when many composers such as Arnold Schoenberg and Richard Strauss were pushing traditional Tonality to, and indeed past, its breaking point, the question of what is music was of particular interest; and Var\èse\’s definition was an eye-opener.\  It solidified an expanded aesthetic point of view that served as both a condition and catalyst for most avant-garde musical work and thought thereafter.\  It is from this definition that most of my thought on the matter has stemmed.\  In considering it together with others and my own experiences in music listening, performance, study and creation, I propose the following adjusted definition: that music is the artful parsing of memory by sound, or as is the case with acoustic music, by performed sound.\ 

\ \ \  The same openness and interpretability of Var\èse\’s definition that allowed for some of the most expansive and valuable contributions to music in the last century now also contributes to the overly-subjective environment of the post-Cagean jungle.\  In order to formulate discussion of a qualitative nature with regard to music there needs to be a more appropriate criterion than the degree to which a piece of music is organized.\ 
\ \ \  My proposed solution to this is perhaps a paradoxical one, and potentially invites even greater degrees of subjectivity.\  After all, it is art\’s very subjective nature that gives it such power and sway over the hearts and minds of those who receive it, and its capacity to provoke thought \— and music is no exception.\  The term \“artful\” offers a channel through which the subjectivity in music is guided and contextualized where it has hitherto been unbounded.\  Though easily as implicit as it is ambiguous, it is with this element of music that I seek to explicitly modify Var\èse\’s use of, \‘organization.\’
\ \ \  In music one is always comparing what one hears in the present to what one has heard in the past and the trained listener will apply learned structure(s) to what he or she is hearing and thereby devise an interpretation.\  This foregrounding of \“memory,\” in combination with the term \“artful,\” elucidates a point that, though only implicit in the definition itself, is perhaps its most important feature: that music requires human agents and interpreters to be.

\ \ \  I have approached my composition, \<b\>Skeleton\</b\>, with these distinctions at the front of my mind.\  The meaning of the title, and the subsequent philosophy of the piece, is therefore twofold.\  \<b\>Skeleton\</b\> simultaneously refers to the foundational function of the score to the music and the music to the memory, and to the ever-fleeting infinitesimal nature of the present moment.\  It alludes not only to the musical and social structures that support it as a \‘piece of music,\’ but also to the decaying remains of the present as it passes through our memory of it.\ 
\ \ \  The piece is made up of many short segments or quanta that are constantly developing or repeating in order to directly address the memory of the present as it flows through the mind the listener.\  The rate at which this flow occurs is also of interest to me, and methods of its control, varying from the proportioning of silences to the fluctuating of tempo to the point of extreme rubato, are employed therein. The quanta\’s eventual combinations and interactions produce increasingly larger chunks of time to the point where their identities are indistinct from one another and little of their origins remain other than the inevitable silence that follows them.\  In order to take full advantage of the human agents involved in the piece\’s performance, the quanta are designed to be fundamental units of human expression (i.e. a gentle or melancholy \‘sighing\’ motive, or clamorous outburst of anger).\  This directly ties the identity and humanity of the performers to the musical output of the piece.

\ \ \  \<b\>Skeleton\</b\> will be premiered in November of 2008, and with a total of six performances across Canada in the same month, I hope to research and judge the effectiveness or ineffectiveness of these thoughts and approaches on both the performed result and its reception with diverse sets of listeners, through direct observations and discussions with the musicians and audiences.\ 
\ \ \  Does this approach affect the resulting music in a way that justifies its use?\  What can be learned, in either case, about the piece, about psychoacoustics and memory, and about musical art in general?}, keywords = {Compositions, Ensemble Contemporain de Montr{\'e}al +, G{\'e}n{\'e}ration 2008, Skeleton}, author = {Michael Berger} } @conference { chafejaffeicassp86, title = {Source Separation and Note Identification in Polyphonic Music}, booktitle = {Proceedings of the International Conference on Acoustics, Speech, and Signal Processing, Tokyo}, number = {STAN-M-34}, year = {1986}, publisher = {IEEE Press}, organization = {IEEE Press}, address = {New York}, keywords = {note identification, polyphonic music, source separation}, url = {https://ccrma.stanford.edu/files/papers/stanm34.pdf}, author = {Chafe, Chris and Jaffe, David A.} } @conference {3656, title = {Source Separation of Polyphonic Music with Interactive User-Feedback on a Piano-Roll Display}, booktitle = {International Society for Music Information Retrieval Conference}, year = {2013}, note = {\ }, month = {11/2013}, address = {Curitiba}, abstract = {\ }, keywords = {Interaction, piano-roll, source separation, user-feedack}, author = {Nicholas Bryan and Mysore, Gautham J. and Ge Wang} } @conference {4706, title = {Speaker and Noise Independent Online Single Channel Speech Enhancement}, booktitle = {2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP)}, year = {2015}, note = {\ }, month = {04/2015}, publisher = {IEEE}, organization = {IEEE}, address = {Brisbane, Australia}, abstract = {Desirable properties of real-world speech enhancement methods include online operation, single-channel operation, operation in the presence of a variety of noise types including non-stationary noise, and no requirement for isolated training examples of the specific speaker and noise type at hand. Methods in the literature typically possess only a subset of these properties. Source separation methods particularly rarely simultaneously possess the first and last properties. We extend universal speech model-based speech enhancement to adaptively learn a noise model in an online fashion. We learn a model from a general corpus of speech in place of speaker-dependent training examples before deployment. This setup provides all of these desirable properties, making it easy to deploy in real-world systems without the need to provide additional training examples, while explicitly modeling speech. Our experimental results show that our method achieves the same performance as in the case in which speaker-dependent training data is available.}, isbn = {978-1-4673-6997-8}, doi = {10.1109/ICASSP.2015.7177934}, url = {https://ieeexplore.ieee.org/document/7177934/}, author = {Fran{\c c}ois G. Germain and Mysore, Gautham J.} } @conference {2588, title = {Stanford Laptop Orchestra (SLOrk)}, booktitle = {International Computer Music Conference}, year = {2009}, month = {08/2009}, address = {Montreal}, abstract = {

In the paper, we chronicle the instantiation and adventures\ of the Stanford Laptop Orchestra (SLOrk), an ensemble of\ laptops, humans, hemispherical speaker arrays, interfaces,\ and, more recently, mobile smart phones. Motivated to\ deeply explore computer-mediated live performance, SLOrk provides a platform for research, instrument design,\ sound design, new paradigms for composition, and\ performance. \ It also offers a unique classroom combining\ music, technology, and live performance. Founded in\ 2008, SLOrk was built from the ground-up by faculty and\ students at Stanford University{\textquoteright}s Center for Computer\ Research in Music and Acoustics (CCRMA). This\ document describes 1) how the founding members built the\ Stanford Laptop Orchestra from laptops, IKEA salad\ bowls, amplifier kits, car speakers, meditation mats and\ pillows, commodity input devices, and custom software, 2)\ the initial performances of SLOrk, including a large-scale\ outdoor concert, as well as a networked concert between\ Stanford and Beijing, and 3) the Stanford Laptop Orchestra\ as a classroom. \ In doing so, we hope to present a history\ of the laptop orchestra, and look to its future.

}, keywords = {ccrma, slork, stanford laptop orchestra}, url = {http://ccrma.stanford.edu/~ge/publish/slork-icmc2009.pdf}, author = {Ge Wang and Nicholas Bryan and Jieun Oh and Rob Hamilton} } @conference {2575, title = {Surviving on Planet CCRMA, two years later and still alive}, booktitle = {International Computer Music Conference}, year = {2005}, month = {09/2005}, address = {Barcelona, Spain}, abstract = {

Planet CCRMA at Home is a collection of packages that you can add to a computer running RedHat 9 or Fedora Core 1, 2 or 3 to transform it into an audio workstation with a lowlatency kernel, current ALSA audio drivers and a nice set of music, midi, audio and video applications. This presentation will outline the changes that have happened in the Planet over the past two years (since the previous presentation at LAC2003).

}, url = {https://ccrma.stanford.edu/~nando/publications/planetccrma_icmc2005.pdf}, author = {Lopez-Lezcano, Fernando} } @conference {2570, title = {The Switched Convolution Reverberator}, booktitle = {127th Audio Engineering Society Convention }, year = {2009}, month = {10/2009}, address = {New York, NY, USA}, abstract = {

An artificial reverberator having low memory and small computational cost, appropriate for mobile devices, is presented. The reverberator consists of an equalized comb filter driving a convolution with a short noise sequence. The reverberator equalization and decay rate are controlled by low-order IIR filters, and the echo density is that of the noise sequence. While this structure is efficient and readily generates high echo densities, if a fixed noise sequence is used, the reverberator has an unwanted periodicity at the comb filter delay length. To overcome this difficulty, the noise sequence is regularly updated or \“switched\”. Several structures for updating the noise sequence, including a leaky integrator sensitive to the signal crest factor, and a multi-band architecture, are described.

}, keywords = {artificial reverberator}, author = {Lee, Keun Sup and Jonathan Abel and Vesa V{\"a}lim{\"a}ki and David P. Berners} } @conference { berdahlandsmithnime08, title = {A Tangible Virtual String}, booktitle = {Proc. 2008 Conf. on New Instruments for Musical Expression (NIME-08), Geneva, Italy}, year = {2008}, author = {Edgar Berdahl and Julius O. Smith} } @conference { chafeetallicmc85, title = {Techniques for Note Identification in Polyphonic Music}, booktitle = {Proceddings of the 1985 International Computer Music Conference, Burnaby, B.C., Canada}, number = {STAN-M-29}, year = {1985}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, keywords = {note identification, polyphonic music}, url = {https://ccrma.stanford.edu/files/papers/stanm29.pdf}, author = {Chafe, Chris and Jaffe, David A. and Kyle Kashima and Mont-Reynaud, Bernard and Julius O. Smith} } @conference {3833, title = {The TR-808 Cymbal: a Physically-Informed, Circuit-Bendable, Digital Model}, booktitle = {40th International Computer Music Conference (ICMC) / 11th Sound and Music Computing Conference (SMC)}, year = {2014}, note = {\ }, month = {09/2014}, address = {Athens, Greece}, abstract = {\ }, keywords = {cymbal, drum machine, physical modeling, TR-808, virtual analog}, url = {http://icmc14-smc14.net/images/proceedings/OS24-B10-TheTR-808Cymbal.pdf}, author = {Kurt James Werner and Abel, Jonathan S and Julius O. Smith} } @conference {3481, title = {TweetDreams: Making music with the audience and the world using real-time Twitter data}, booktitle = { International Conference on New Interfaces For Musical Expression}, year = {2011}, month = {05/2011}, address = {Oslo, Norway}, abstract = {TweetDreams is an instrument and musical composition which creates real-time sonification and visualization of tweets. Tweet data containing specified search terms is retrieved from Twitter and used to build networks of associated tweets. These networks govern the creation of melodies associated with each tweet and are displayed graphically. Audience members participate in the piece by tweeting, and their tweets are given special musical and visual prominence.

}, author = {Dahl, Luke and Jorge Herrera and Wilkerson, Carr} } @conference {3476, title = {Two Turntables and a Mobile Phone}, booktitle = {International Conference on New Interfaces for Musical Expression}, year = {2011}, note = {\ }, month = {05/2011}, address = {Oslo, Norway}, abstract = {\ }, author = {Nicholas Bryan and Ge Wang} } @conference {4704, title = {Uniform Noise Sequences for Nonlinear System Identification}, booktitle = {15th International Conference on Digital Audio Effects (DAFx-12)}, year = {2012}, note = {\ }, month = {09/2012}, address = {York, UK}, abstract = {Noise-based nonlinear system identification techniques using Hammerstein and Wiener forms have found wide application in biological system modeling, and been applied to modeling nonlinear audio processors such as the ring modulator. These methods apply noise to the system, and project the system output onto a set of orthogonal polynomials to reveal parameters of the model. Though Gaussian sequences are invariably used to drive the unknown system, it seems clear that the statistics of the input will affect the model estimate. Motivated by the limited input and output ranges supported by analog systems, in this work, the use of an input noise sequence having a uniform distribution is explored. In addition, an error measure indicating harmonic distortion modeling accuracy is introduced. Simulation results identifying Hammerstein and Wiener systems show that the uniform and Gaussian distributions perform differently, with the uniform distribution generally producing more accurate harmonic responses. Finally, uniform noise and Gaussian noise are used to model a saturating low-pass circuit similar to that of the Tube Screamer, with the uniform distribution providing a modest improvement in noise response error.}, url = {https://www.dafx12.york.ac.uk/papers/dafx12_submission_79.pdf}, author = {Fran{\c c}ois G. Germain and Abel, Jonathan S and Philippe Depalle and Marcelo M. Wanderley} } @conference {3478, title = {User-Guided Variable-Rate Time-Stretching Via Stiffness Control}, booktitle = {15th International Conference on Digital Audio Effects}, year = {2012}, note = {\ }, month = {09/2012}, address = {York, United Kingdom}, abstract = {\ }, author = {Nicholas Bryan and Jorge Herrera and Ge Wang} } @conference { smithgenhamming, title = {Virtual Electric Guitars and Effects Using Faust and Octave}, booktitle = {Proc. 6th Int. Linux Audio Conf. (LAC2008), http://lac.linuxaudio.org/}, volume = {33}, number = {3}, year = {2008}, note = {paper: http://ccrma.stanford.edu/realsimple/faust\_strings/faust\_strings.pdf} overheads: http://ccrma-ftp.stanford.edu/\ jos/pdf/LAC2008-jos.pdf} supporting website: http://ccrma.stanford.edu/realsimple/faust\_strings/}, month = {Winter}, pages = {1{\textendash}10}, publisher = {Systems Control Tech.}, organization = {Systems Control Tech.}, address = {Toronto, Canada}, author = {Julius O. Smith}, editor = {Jim Beauchamp} } @conference {4125, title = {Wave Digital Filter Adaptors for Arbitrary Topologies and Multiport Linear Elements}, booktitle = {International Conference on Digital Audio Effects (DAFx-15)}, year = {2015}, note = {\ }, month = {November/2015}, address = {Trondheim, Norway}, abstract = {\ }, url = {http://www.ntnu.edu/web/dafx15/program}, author = {Kurt James Werner and Smith III, Julius O. and Abel, Jonathan S} } @conference {4341, title = {Wave Digital Filter Modeling of Circuits with Operational Amplifiers}, booktitle = {European Signal Processing Conference (EUSIPCO)}, year = {2016}, note = {\ }, month = {08/2016}, address = {Budapest, Hungary}, abstract = {\ }, url = {http://www.eusipco2016.org/en}, author = {Kurt James Werner and W. Ross Dunkel and Maximilian Rest and Michael J. Olsen and Smith III, Julius O.} } @conference { huangserafinandsmithicmc00, title = {A Waveguide Mesh Model of High-Frequency Violin Body Resonances}, booktitle = {Proc. 2000 Int. Computer Music Conf., Berlin}, year = {2000}, month = {aug}, author = {Patty Huang and Stefania Serafin and Julius O. Smith} } @conference { abelhuangandsmithaes07, title = {Waveguide Mesh Reverberator with Internal Decay and Diffusion Structures}, booktitle = {Proceedings of the 123rd Convention of the Audio Engineering Society}, year = {2007}, publisher = {preprint 7194}, organization = {preprint 7194}, address = {New York}, author = {Jonathan Abel and Patty Huang and Julius O. Smith} } @conference {4117, title = {The XQIFFT: Increasing the Accuracy of Quadratic Interpolation of Spectral Peaks via Exponential Magnitude Spectrum Weighting}, booktitle = {International Computer Music Conference (ICMC)}, year = {2015}, note = {\ }, month = {September/2015}, address = {Denton, Texas}, abstract = {\ }, author = {Kurt James Werner} } @conference { caceres2009:jacktrip-CMJStyle, title = {{J}ack{T}rip: Under the Hood of an Engine for Network Audio}, booktitle = {Proceedings of International Computer Music Conference}, year = {2009}, pages = {509{\textendash}512}, publisher = {International Computer Music Association}, organization = {International Computer Music Association}, address = {San Francisco, California}, author = {C{\'a}ceres, Juan-Pablo and Chafe, Chris} } @conference {2580, title = {{\textquotedblleft}Separation by Humming{\textquotedblright}: User Guided Sound Extraction from Monophonic Mixtures}, booktitle = {IEEE Workshop on Applications of Signal Processing to Audio and Acoustics (WASPAA)}, year = {2009}, abstract = {

In this paper we present a novel approach for isolating and removing sounds from dense monophonic mixtures. The approach is user-based, and requires the presentation of a guide sound that mimics the desired target the user wishes to extract. The guide sound can be simply be produced from a user by vocalizing or otherwise replicating the target sound marked for separation. Using that guide as a prior in a statistical sound mixtures model, we propose a methodology that allows us to efficiently extract complex structured sounds from dense mixtures.

}, url = {https://ccrma.stanford.edu/~gautham/Site/Humming.html}, author = {Paris Smaragdis and Mysore, Gautham J.} } @proceedings {3554, title = { A Flexible Slide-In System for Computer-Assisted Musicological Analysis}, journal = {5th World Multiconference on Systemics, Cybernetics and Informatics}, volume = {10}, year = {2001}, note = {\ }, pages = {51-56}, abstract = {\ }, author = {Kornstaedt, A.} } @proceedings {2646, title = {Alias-free virtual analog oscillators using a feedback delay loop}, journal = {the 12th Int. Conference on Digital Audio Effects}, year = {2009}, note = {\ }, month = {09/2009}, address = {Como, Italy}, abstract = {
The rich spectra of classic waveforms (sawtooth, square and triangular)
are obtained by discontinuities in the waveforms or their
derivatives. At the same time, the discontinuities lead to aliasing
when the waveforms are digitally generated. To remove or reduce
the aliasing, researchers have proposed various methods,
mostly based on limiting bandwidth or smoothing the waveforms.
This paper introduces a new approach to generate the virtual
analog oscillators with no aliasing. The approach relies on
generating an impulse train using a feedback delay loop, often
used for the physical modeling of musical instruments. Classic
waveforms are then derived from the impulse train with a leaky
integrator. Although the output generated by this method is not
exactly periodic, it perceptually sounds harmonic. While additional
processing is required for time-varying pitch shifting, resulting
in some high-frequency attenuation when the pitch changes,
the proposed method is computationally more efficient than
other algorithms and the high-frequency attenuation can be also
adjusted.
}, keywords = {anti-aliasing, delay-line, fractional-delay filter, oscillators, virtual analog}, url = {http://dafx09.como.polimi.it/proceedings/papers/paper_72.pdf}, author = {Juhan Nam and Vesa V{\"a}lim{\"a}ki and Jonathan Abel and Julius O. Smith} } @proceedings {2750, title = {An Allpass Approach to Digital Phasing and Flanging}, journal = {International Computer Music Conference}, year = {1984}, address = {Paris}, keywords = {Allpass, Approach, digital, Flanging, Phasing}, url = {https://ccrma.stanford.edu/files/papers/stanm21.pdf}, author = {Julius O. Smith} } @proceedings {2562, title = {Back To The Sea: A Software Realization of Dexter Morrill{\textquoteright}s "Sea Songs"}, journal = {International Computer Music Conference}, year = {2009}, month = {08/30/2007}, address = {Copenhagen, Denmark}, abstract = {

This paper describes the technical and aesthetic challenges faced in the recent software-based recreation of composer Dexter Morrill\’s work Sea Songs for soprano voice, computer-generated tape and Radio-Baton controlled hardware effects-processor. Through careful analysis of the composer\’s own notes as well as through extensive testing of Morrill\’s original Digitech TSR-24 stereo-effects processor, a flexible and extensible software emulation of the piece was created as a Max/MSP application.

}, author = {Rob Hamilton} } @proceedings {3552, title = {CCARH{\textquoteright}s Musical Databases on the Web: A Gold Mine for Musicologists}, journal = {Proceedings of the XII Colloquium on Musical Informatics}, year = {1998}, note = {\ }, pages = {280-281}, abstract = {\ }, author = {Kornstaedt, A.} } @proceedings { ccrmaicmc91, title = {CCRMA Papers on Physical Modeling from the 1991 International Computer Music Conference, Montreal, Canada}, journal = {International Computer Music Conference}, number = {STAN-M-73}, year = {1991}, month = {10/1991}, address = {Montreal, Canada}, keywords = {icmc}, url = {https://ccrma.stanford.edu/files/papers/stanm73.pdf}, author = {Cook, Perry and Hirschman, Suzanne and Julius O. Smith} } @proceedings { ccrmaicmc93, title = {CCRMA Papers Presented at the 1993 International Computer Music Conference, Tokyo, Japan}, journal = {International Computer Music Conference}, number = {STAN-M-81}, year = {1993}, note = {Papers by Chafe, Cook, Morrill, Smith, and Van Duyne.}, month = {12/1993}, address = {Tokyo, Japan}, keywords = {icmc}, url = {https://ccrma.stanford.edu/files/papers/stanm81.pdf}, author = {Chafe, Chris and Cook, Perry and Morrill and Julius O. Smith and van Duyne, Scott} } @proceedings { ccrmaicmc94, title = {CCRMA Papers Presented at the 1994 International Computer Music Conference, Arhus, Denmark}, year = {1994}, note = {Papers by Julius O. Smith, Davide Rochesso, Scott Van Duyne, David Berners, Gary P. Scavone, Perry R. Cook, John R. Pierce, Brent Gillespie, Fernando Lopez-Lezcano, Jan Chomyszyn, Enrique Moreno, Davis A. Jaffe and W. Andrew Schloss.}, month = {09/1994}, keywords = {icmc}, url = {https://ccrma.stanford.edu/files/papers/stanm89.pdf}, author = {Julius O. Smith and Divide Rochesso and van Duyne, Scott and David P. Berners and Scavone, Gary P. and Cook, Perry and Pierce, John R. and Gillespie, Brent and Lopez-Lezcano, Fernando and Jan Chomyszyn and Enrique Moreno and Jaffe, David A. and Schloss, W. A.} } @proceedings {2795, title = {Characterization of Aperiodicity in Nearly Periodic Signals}, journal = {International Conference on Acoustics, Speech and Signal Processing}, number = {STAN-M-62}, year = {1990}, publisher = {IEEE Press}, address = {Albuquerque}, keywords = {aperiodicity, periodic}, url = {https://ccrma.stanford.edu/files/papers/stanm62.pdf}, author = {Schumacher, R. T. and Chafe, Chris} } @proceedings {2680, title = {Compositional Tools for Adding Expression to Music}, journal = {International Computer Music Conference}, number = {STAN-M-82}, year = {1993}, month = {10/1993}, publisher = {CCRMA}, address = {San Jose, CA, USA}, keywords = {Composition, Composition tools, dmix, Expression, music, Music Expression}, url = {https://ccrma.stanford.edu/files/papers/stanm82.pdf}, author = {Oppenheim, Daniel V.} } @proceedings {2681, title = {DMIX: An Environment for Composition}, journal = {International Computer Music Conference}, year = {1989}, month = {05/1989}, publisher = {CCRMA}, address = {Columbus, Ohio, USA}, keywords = {Composition, Composition Environment, dmix}, url = {https://ccrma.stanford.edu/files/papers/stanm60.pdf}, author = {Oppenheim, Daniel V.} } @proceedings { igoudinandsmith96, title = {Doppler Simulation and the Leslie}, journal = {Proc. COST-G6 Conf. Digital Audio Effects (DAFx-02), Hamburg, Germany}, volume = {20}, number = {2}, year = {2002}, note = {crma.stanford.edu/\ jos/doppler/} https://ccrma.stanford.edu/\ jos/doppler/}}, month = {September 26}, pages = {13{\textendash}20}, publisher = {CCRMA, Stanford University}, address = {New York}, author = {Julius O. Smith and Stefania Serafin and Jonathan Abel and David P. Berners}, editor = {Jim Beauchamp} } @proceedings {2752, title = {Efficient Yet Accurate Models for Strings and Air Columns Using Sparse Lumping of Distributed Losses and Dispersion}, journal = {Colloquium on Physical Modeling}, number = {STAN-M-67}, year = {1990}, month = {12/1990}, address = {Grenoble, France}, keywords = {Accurate, Air Columns, Dispersion, Distributed, Efficient, Losses, Lumping, Models, Sparse, Strings}, url = {https://ccrma.stanford.edu/files/papers/stanm67.pdf}, author = {Julius O. Smith} } @proceedings {2813, title = {Elimination of Limit Cycles and Overflow Oscillations in Time-Varying Lattive and Ladder Digital Filters}, journal = {IEEE Conference on Circuits and Systems}, number = {STAN-M-35}, year = {1986}, note = {Full version available in STAN-M-39: https://ccrma.stanford.edu/papers/music-applications-of-digital-waveguides
}, month = {05/1986}, address = {San Jose, CA, USA}, keywords = {digital filters, ladder, lattice, limit cycles, overflow oscillations, time-varying}, url = {https://ccrma.stanford.edu/files/papers/stanm35.pdf}, author = {Julius O. Smith} } @proceedings {2771, title = {Ensemble Timing in Computer Music}, journal = {International Computer Music Conference}, number = {STAN-M-23}, year = {1983}, address = {Eastman School of Music, Rochester, N.Y.}, keywords = {computer music, ensemble timing}, url = {https://ccrma.stanford.edu/files/papers/stanm23.pdf}, author = {Jaffe, David A.} } @proceedings {2564, title = {For Jean-Claude: [re]Presenting Duet for One Pianist}, journal = {International Computer Music Conference}, year = {2009}, month = {08/2009}, address = {Montreal, Canada}, abstract = {

\ In 1989, composer and researcher Jean-Claude Risset{\textquoteright}s\ series of interactive sketches for piano and Disklavier\ entitled Duet for One Pianist explored the performative\ possibilities made available to pianists through the\ augmentation of emotive human musical gesture with the\ precise reactive and computational capabilities afforded by\ computer-based musical systems. As computer and\ musical software systems have evolved, the Max software\ patches created by Risset and researcher Scott Van-Duyne\ at the MIT Media Lab have been updated and maintained\ to allow the pieces to be performed using contemporary\ hardware and software systems. In distinct contrast,\ Risset{\textquoteright}s original hand-notated musical score for the work -\ representing performance notation for the human pianist\ alongside a varying level of detail representing the\ computer\’s response, itself an integral part in the work -\ remains the authoritative representation available to\ performers, researchers and archivists alike. This paper\ outlines ongoing efforts towards the augmentation of\ Risset{\textquoteright}s existing score through the production of a\ comprehensive multi-voiced notated score edition of Duet\ for One Pianist, as well as symbolic and data\ representations for each of the eight works derived from\ live performance data, and a complimentary and complete\ series of audio and visual recordings of Duet by pianist\ Chryssie Nanou.

}, author = {Rob Hamilton and Chryssie Nanou} } @proceedings {4504, title = {Game Design for Expressive Mobile Music}, journal = {New Interfaces for Musical Expression}, year = {2016}, note = {\ }, month = {07/2016}, address = {Brisbane, Australia}, abstract = {This article presents observations and strategies for designing game- like elements for expressive mobile musical interactions. The designs of several popular commercial mobile music instruments are discussed and compared, along with the different ways they integrate musical information and game-like elements. In particular, issues of designing goals, rules, and interactions are balanced with articulating expressiveness. These experiences aim to invite and engage users with game design while maintaining and encouraging open-ended musical expression and exploration. A set of observations is derived, leading to a broader design motivation and philosophy.}, keywords = {game design, gamification, mobile music, musical expression}, author = {Ge Wang} } @proceedings {2756, title = {Hardware, Software, and Compositional Tools for a Real Time Improvised Solo Trumpet Work}, journal = {International Computer Music Conference}, number = {STAN-M-56}, year = {1989}, address = {Columbus, Ohio, USA}, keywords = {Composition, hardware, icmc, improvised, real time, software, trumpet}, url = {https://ccrma.stanford.edu/files/papers/stanm56.pdf}, author = {Cook, Perry} } @proceedings {3555, title = {The JRing System for Computer-Assisted Musicological Analysis}, journal = { 2nd Annual International Symposium on Music Information Retrieval}, year = {2001}, note = {\ }, pages = {93-98}, abstract = {\ }, doi = {10.1.1.15.9532}, author = {Kornstaedt, A.} } @proceedings {4507, title = {The Laptop Accordion}, journal = {New Interfaces for Musical Expression}, year = {2016}, note = {\ }, month = {07/2016}, address = {Brisbane, Australia}, abstract = {The{\textquotedblleft}Laptop Accordion{\textquotedblright}co-opts the commodity laptop computer to craft an expressive, whimsical accordion-like instrument. It utilizes the opening and closing of the laptop screen as a physical metaphor for accordion bellows, and the laptop keyboard as a musical buttonboard. Motion is tracked using the laptop camera via optical flow and mapped to continuous control over dynamics, while the sound is generated in real-time. The instrument uses both skeuomorphic and abstract onscreen graphics which further reinforce the core mechanics of {\textquotedblleft}squeezebox{\textquotedblright} instruments. The Laptop Accordion provides several game modes, while overall offering an unconventional aesthetic experience in music making.}, keywords = {design, laptop accordion}, author = {Aidan Meacham and Sanjay Kannan and Ge Wang} } @proceedings {2747, title = {LECTOR: An Ecclesiastical Latin Control Language for the SPASM/Singer Instrument}, journal = {International Computer Music Conference}, number = {STAN-M-76}, year = {1991}, address = {Montreal, Canada}, keywords = {control language, LECTOR, singer instrument, SPASM}, url = {https://ccrma.stanford.edu/files/papers/stanm76.pdf}, author = {Cook, Perry} } @proceedings {3450, title = {Linux Audio Conference 2012 Proceedings}, journal = {Linux Audio Conference 2012}, year = {2012}, month = {04/2012}, publisher = {CCRMA, Stanford University}, edition = {2012}, address = {CCRMA, Stanford University}, isbn = {978-1-105-62546-6}, url = {http://lac.linuxaudio.org/2012/download/lac2012_proceedings.pdf}, author = {Various Author}, editor = {Neuman, Frank and Lopez-Lezcano, Fernando} } @proceedings {2566, title = {maps and legends: FPS-Based Interfaces For Composition and Immersive Performance}, journal = {International Computer Music Conference}, year = {2007}, address = {Copenhagen, Denmark}, abstract = {

This paper describes an interactive multi-channel multiuser networked system for real-time composition and immersive performance built using a modified version of the Quake III gaming engine. By tracking users positional and action data within a virtual space, and by streaming that data over a network using OSC messages formatted as UDP packets to a multi-channel Pure Data patch, actions in virtual space are correlated to sonic output in a physical space. Virtual environments designed as abstract compositional maps or representative models of the users actual physical space are investigated as means to guide and shape compositional and performance choices. This paper analyzes both the technological concerns for building and realizing the system as well as the compositional and perceptual issues inherent in the project itself.

}, keywords = {virtual worlds}, author = {Rob Hamilton} } @proceedings {4675, title = {Network Variable Preserving Step-size Control In Wave Digital Filters}, journal = {20th International Conference on Digital Audio Effects (DAFx-17)}, year = {2017}, note = {\ }, month = {09/2017}, pages = {200{\textendash}207}, address = {Edinburgh, United Kingdom}, abstract = {\ }, author = {Olsen, Michael J{\o}rgen and Kurt James Werner and Fran{\c c}ois G. Germain} } @proceedings {2677, title = {A New Approach to Digital Reverberation Using Closed Waveguide Networks}, journal = {International Computer Music Conference}, number = {STAN-M-31}, year = {1985}, month = {1985}, pages = {47-53}, address = {Burnaby, B.C., Canada}, keywords = {closed waveguide networks, digital reverberation, reverberation}, url = {https://ccrma.stanford.edu/files/papers/stanm31.pdf}, author = {Julius O. Smith} } @proceedings {2760, title = {NUTATION: Structural Organization Versus Graphical Generality in a Common Music}, journal = {International Computer Music Conference}, number = {STAN-M-59}, year = {1989}, month = {09/1989}, address = {Columbus, Ohio, USA}, keywords = {common music, graphical generality, notation, NUTATION, structural organization}, url = {https://ccrma.stanford.edu/files/papers/stanm59.pdf}, author = {Diener, G.} } @proceedings {2676, title = {On Finding Rythmic Patterns in Musical Lines}, journal = {ICMC {\textquoteright}85}, number = {STAN-M-30}, year = {1985}, month = {07/1985}, publisher = {CCRMA}, keywords = {Musical Patterns, Patterns, Rhythmic, Rhythmic Patterns}, url = {https://ccrma.stanford.edu/files/papers/stanm30.pdf}, author = {Mont-Reynaud, Bernard and Goldstein, Mark} } @proceedings {2800, title = {PARSHL: A Program for the Analysis/Synthesis of Inharmonic Sounds Based on a Sinusoidal Representation}, journal = {International Computer Music Conference}, number = {STAN-M-43}, year = {1987}, address = {Champaign-Urbana, Illinois}, keywords = {analysis, inharmonic sounds, PARSHL, sinusoidal representation, synthesis}, url = {https://ccrma.stanford.edu/files/papers/stanm43.pdf}, author = {Julius O. Smith and Serra, Xavier} } @proceedings { ccrmaicmc92, title = {Physical Modeling and Signal Processing: CCRMA Papers Presented at the 1992 International Computer Music Conference, San Jose, CA}, number = {STAN-M-80}, year = {1992}, note = {Papers by Cook, Gillespie, Smith, Tanaka, and Van Duyne.}, month = {oct}, keywords = {icmc, physical modeling, signal processing}, url = {http://ccrma.stanford.edu/files/papers/stanm80.pdf}, author = {Cook, Perry and Gillespie, Brent and Julius O. Smith and Tanaka and van Duyne, Scott} } @proceedings {2642, title = {A physically motivated room reverberation enhancement system that is stable in any (dissipative) room: An application of sound portholes}, journal = {NOISE-CON 2010 in conjunction ith the 159th Meeting of the Acoustical Society of America}, year = {2010}, month = {04/2010}, address = {Baltimore, MD}, abstract = {Prior active room reverberation enhancement systems have typically employed
microphones, artificial reverberator filters, and loudspeakers to change the reverberant
properties of a room. However, acoustic feedback from the loudspeakers to the
microphones has had the potential to drive such systems unstable. To avoid feedback
instabilities, we apply passivity techniques from control theory to design a stable room
reverberation enhancement system from the ground up. In particular, we employ sound
portholes, which are special transducers that operate concurrently as microphones and
loudspeakers.\  With these sound portholes, the feedback controller implements a passive
connection to virtual acoustic spaces, which are realized using digital waveguide networks.\ 
Because the feedback controller models a passive system, it is theoretically stable in any
(dissipative) acoustic environment and for arbitrarily large loop gains. As a consequence,
the system does not suffer from \“ringing tones\” at high loop gains in the same way as prior
systems have suffered. Furthermore, the system does not need to be re-calibrated if the
properties of the room change or even if moved to a whole new room. This method for
designing room reverberation enhancement systems may generally result in more realistic
reverberant sound because it implements the acoustical features of a system that could
exist naturally in the physical world.
}, url = {https://ccrma.stanford.edu/~eberdahl/Papers/NOISECON2010Reverb.pdf}, author = {Edgar Berdahl and Gunter Niemeyer and Julius O. Smith} } @proceedings { caceres2008:playing, title = {Playing the Network: the Use of Time Delays as Musical Devices}, journal = {International Computer Music Conference}, year = {2008}, pages = {244-250}, author = {C{\'a}ceres, Juan-Pablo and Renaud, Alain B.} } @proceedings {2688, title = {Proceedings of the 15th International Congress on Acoustics}, journal = {International Congress of Acoustics}, year = {1995}, month = {06/1995}, publisher = {CCRMA}, address = {Trondheim, Norway}, keywords = {International Congress}, author = {Julius O. Smith} } @proceedings {2567, title = {q3osc: or How I Learned To Stop Worrying and Love The Game}, journal = {International Computer Music Conference}, year = {2008}, address = {Belfast, Ireland}, abstract = {

\ q3osc is a heavily modified version of the open-sourced ioquake3 gaming engine featuring an integrated Oscpack implementation of Open Sound Control for bi-directional communication between a game server and one or more external audio servers. By combining ioquake3\’s internal physics engine and robust multiplayer network code with a full-featured OSC packet manipulation library, the virtual actions and motions of game clients and previously one-dimensional in-game weapon projectiles can be repurposed as independent and behavior-driven OSC emitting sound-objects for real-time networked performance and spatialization within a multi-channel audio environment. This paper details the technical and aesthetic decisions made during the development and initial implementations of q3osc and introduces specific mapping and spatialization paradigms currently in use for sonification.

}, keywords = {q3osc, virtual worlds}, author = {Rob Hamilton} } @proceedings { caceres2005:scuba, title = {SCUBA: the Self-Contained Unified Bass Augmenter}, journal = {New Interfaces for Musical Expression}, year = {2005}, pages = {38{\textendash}41}, author = {C{\'a}ceres, Juan-Pablo and Mysore, Gautham J. and Jeffrey Trevi{\~n}o} } @proceedings {2678, title = {Slappability: A New Metaphor for Human Computer Interaction}, journal = {World Conference on Artificial Intelligence}, year = {1993}, month = {10/1993}, publisher = {CCRMA}, address = {Stanford, CA}, keywords = {computer, dmix, Human, Human Computer Interaction, Interaction, music objects, Slappability}, url = {https://ccrma.stanford.edu/files/papers/stanm85.pdf}, author = {Oppenheim, Daniel V.} } @proceedings {2745, title = {Synthesis of the Singing Voice Using a Physically Parameterized Model of the Human Vocal Tract}, journal = {International Computer Music Conference}, number = {STAN-M-57}, year = {1989}, address = {Columbus, Ohio, USA}, keywords = {human vocal tract, physical model, singing voice, synthesis}, url = {https://ccrma.stanford.edu/files/papers/stanm57.pdf}, author = {Cook, Perry} } @proceedings { caceres2008:china, title = {To the Edge with China: Explorations in Network Performance}, journal = {ARTECH 2008: Proceedings of the 4th International Conference on Digital Arts}, year = {2008}, pages = {61{\textendash}66}, address = {Porto, Portugal}, isbn = {978-989-95776-3-3}, author = {C{\'a}ceres, Juan-Pablo and Rob Hamilton and Iyer, Deepak and Chafe, Chris and Ge Wang} } @proceedings {2650, title = {Variable Fractional Delay Filters in Bandlimited Oscillator Algorithms for Music Synthesis}, journal = {the 2010 International Conference on Green Circuits and Systems (ICGCS2010)}, year = {2010}, note = {\ }, month = {06/2010}, address = {Shanghai, China}, abstract = {Trivially sampled geometric waveforms such as the\ 
rectangular pulse wave used in subtractive sound synthesis suffer\ 
from aliasing caused by the discontinuities in the waveform or its\ 
derivative. Several algorithms for the reduction of aliasing dis-\ 
tortion have been suggested, providing either complete removal\ 
or great suppression of aliasing. Some antialiasing oscillators\ 
utilize variable fractional delay filters as an essential part of\ 
the algorithm. In this paper, these oscillators are reviewed with\ 
an emphasis on motivating the use of the fractional delay filters.
}, keywords = {Antialiasing, fractional-delay filter, oscillators, virtual analog}, author = {Jussi Pekonen and Vesa V{\"a}lim{\"a}ki and Juhan Nam and Jonathan Abel and Julius O. Smith} } @article {2649, title = {Alias-suppressed oscillators based on differentiated polynomial waveforms}, journal = {IEEE Transactions on Audio, Speech and Language Processing}, volume = {18}, year = {2010}, note = {\ }, month = {05/2010}, chapter = {786}, abstract = {\ An efficient approach to the generation of classical synthesizer waveforms with reduced aliasing is proposed. This paper introduces two new classes of polynomial waveforms that can be differentiated one or more times to obtain an improved version of the sampled sawtooth and triangular signals. The differentiated polynomial waveforms (DPW) extend the previous differentiated parabolic wave method to higher polynomial orders, providing improved alias-suppression. Suitable polynomials of order higher than two can be derived either by analytically integrating a previous lower order polynomial or by solving the polynomial coefficients directly from a set of equations based on constraints. We also show how rectangular waveforms can be easily produced by differentiating a triangular signal. Bandlimited impulse trains can be obtained by differentiating the sawtooth or the rectangular signal. An objective evaluation using masking and hearing threshold models shows that a fourth-order DPW method is perceptually alias-free over the whole register of the grand piano. The proposed methods are applicable in digital implementations of subtractive sound synthesis.}, keywords = {acoustic signal processing, Antialiasing, audio oscillators, music, signal synthesis}, issn = {1558-7916}, doi = {10.1109/TASL.2009.2026507}, url = {http://www.ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5153306}, author = {Vesa V{\"a}lim{\"a}ki and Juhan Nam and Julius O. Smith and Jonathan Abel} } @article {2668, title = {Analysis of pitch perception of iharmonicity in pipa strings using response surface methodology}, journal = {Journal of New Music Research}, volume = {39}, year = {2010}, month = {05/2010}, type = {computational musicology}, chapter = {63}, url = {http://www.informaworld.com/smpp/title~db=all~content=g921838317}, author = {Shin Hui Lin Chin and Jonathan Berger} } @article { georgesmith92, title = {Analysis-by-Synthesis/Overlap-add Sinusoidal Modeling Applied to the Analysis and Synthesis of Musical Tones}, journal = {J. Audio Eng. Soc.}, volume = {40}, number = {6}, year = {1992}, pages = {497{\textendash}516}, author = {E. B. George and M. J. T. Smith} } @article {2573, title = {Audio Watermarking through Deterministic plus Stochastic Signal Decomposition}, journal = {EURASIP Journal on Information Security}, year = {2007}, pages = {Volume 2007, Article ID 75961, 12 pages}, abstract = {

\ This paper describes an audio watermarking scheme based on sinusoidal signal modeling. To embed a watermark in an original\ signal (referred to as a cover signal hereafter), the following steps are taken. (a) A short-time Fourier transform is applied to the\ cover signal. (b) Prominent spectral peaks are identified and removed. (c) Their frequencies are subjected to quantization index\ modulation. (d) Quantized spectral peaks are added back to the spectrum. (e) Inverse Fourier transform and overlap-adding\ produce a watermarked signal. To decode the watermark, frequencies of prominent spectral peaks are estimated by quadratic\ interpolation on themagnitude spectrum. Afterwards, a maximum-likelihood procedure determines the binary value embedded in\ each frame. Results of testing against lossy compression, low- and highpass filtering, reverberation, and stereo-to-mono reduction\ are reported. A Hamming code is adopted to reduce the bit error rate (BER), and ways to improve sound quality are suggested as\ future research directions.

}, keywords = {sinusoidal modeling, Watermarking}, url = {http://www.hindawi.com/journals/is/2007/075961.abs.html}, author = {Liu, Yi-Wen and Julius O. Smith} } @article {4500, title = {ChucK: A Strongly-timed Computer Music Language}, journal = {Computer Music Journal}, volume = {39}, year = {2015}, note = {\ }, month = {12/2015}, type = {design, engineering; domain specific programming language}, chapter = {10-29}, abstract = {ChucK is a programming language designed for computer music. It aims to be expressive and straightforward to read and write with respect to time and concurrency, and to provide a platform for precise audio synthesis and analysis and for rapid experimentation in computer music. In particular, ChucK defines the notion of a strongly timed audio programming language, comprising a versatile time-based programming model that allows programmers to flexibly and precisely control the flow of time in code and use the keyword now as a time-aware control construct, and gives programmers the ability to use the timing mechanism to realize sample-accurate concurrent programming. Several case studies are presented that illustrate the workings, properties, and personality of the language. We also discuss applications of ChucK in laptop orchestras, computer music pedagogy, and mobile music instruments. Properties and affordances of the language and its future directions are outlined.}, keywords = {audio, ChucK, concurrent, design, music, on-the-fly programming, programming language, strongly-timed}, doi = {10.1162/COMJ_a_00324}, url = {http://chuck.stanford.edu/}, author = {Ge Wang and Perry R. Cook and Spencer Salazar} } @article {2690, title = {Common Music: A Music Composition Language in Common Lisp and CLOS}, journal = {Computer Music Journal}, volume = {15}, year = {1991}, month = {07/1991}, chapter = {21}, keywords = {CLOS, Composition, Composition Language, Lisp}, url = {http://www.jstor.org/stable/3680913}, author = {Taube, Heinrich} } @article {2579, title = {Composing for Laptop Orchestra}, journal = {Computer Music Journal}, year = {2008}, pages = {32(1):9-25}, abstract = {

The authors describe in detail several pieces composed in this past year for the Princeton Laptop Orchestra (PLOrk). Particular attention is paid to sonic strategies, networking and the conductor, interface design, and game pieces. This paper should function as both a record of the first pieces composed for PLOrk and a guide for future laptop orchestra composers.

}, keywords = {composing, laptop orchestra, plork, princeton}, author = {Smallwood, Scott and Dan Trueman and Cook, Perry and Ge Wang} } @article {2784, title = {The Computer as a Musical Instrument}, journal = {Scientific American}, volume = {256}, number = {STAN-M-37}, year = {1987}, chapter = {90}, keywords = {computer, music}, url = {https://ccrma.stanford.edu/files/papers/stanm37.pdf}, author = {Mathews, Max Vernon and Pierce, John R.} } @article { abeandsmith04, title = {Design Criteria for Simple Sinusoidal Parameter Estimation based on Quadratic Interpolation of FFT Magnitude Peaks}, journal = {Audio Engineering Society Convention, San Francisco}, year = {2004}, note = {Preprint 6256}, publisher = {Audio Engineering Society}, address = {New York}, author = {Mototsugu Abe and Julius O. Smith} } @article {2648, title = {Efficient antialiasing oscillator algorithms using low-order fractional delay filters}, journal = {IEEE Transactions on Audio, Speech and Language Processing}, volume = {18}, year = {2010}, note = {\ }, month = {05/2010}, chapter = {773}, abstract = {\ One of the challenges in virtual analog synthesis is avoiding aliasing when generating classic waveforms such as sawtooth and square wave which have theoretically infinite bandwidth in their ideal forms. The human auditory system renders a certain amount of aliasing inaudible, which allows room for finding cost-effective algorithms. This paper suggests efficient algorithms to reduce the aliasing using low-order fractional delay filters in the framework of bandlimited impulse train (BLIT) synthesis. Examining Lagrange, B-spline interpolators and allpass fractional delay filters, optimized methods will be discussed for generating classic waveforms (sawtooth, square, and triangle). Techniques for generating more complicated harmonics such as pulse width modulation, hard-sync, and super-saw are also presented. The perceptual evaluation is performed by comparing the threshold of hearing and masking curve of oscillators with their aliasing levels. The result shows that the BLIT using the computationally efficient third-order B-spline generates waveforms that are perceptually free of aliasing within practically used fundamental frequencies.}, keywords = {Antialiasing, audio oscillators, audio signal processing, interpolation, signal synthesis}, issn = {1558-7916}, doi = {10.1109/TASL.2009.2035039}, url = {http://www.ieeexplore.ieee.org/xpls/abs_all.jsp?arnumber=5299036\&tag=1}, author = {Juhan Nam and Vesa V{\"a}lim{\"a}ki and Jonathan Abel and Julius O. Smith} } @article { bilbaoandsmithacustica05, title = {Energy-conserving Finite Difference Schemes for Nonlinear Strings}, journal = {Acustica{\textendash}-Acta Acustica}, volume = {91(2)}, year = {2005}, pages = {299{\textendash}311}, author = {Stefan Bilbao and Julius O. Smith} } @article { bilbaoandsmithjasp04-bogus-right?, title = {Energy-conserving Finite Difference Schemes for Nonlinear Strings}, journal = {EURASIP Journal on Applied Signal Processing}, volume = {4}, year = {2004}, note = {(special issue on string modeling), in review}, author = {Stefan Bilbao and Julius O. Smith} } @article {berdahl2012feedback, title = {Feedback control of acoustic musical instruments: Collocated control using physical analogs}, journal = {The Journal of the Acoustical Society of America}, volume = {131}, year = {2012}, note = {\ }, pages = {963}, abstract = {\ }, author = {Berdahl, E. and Smith III, J.O. and Niemeyer, G.} } @article { bilbaoandsmith2003, title = {Finite difference schemes and digital waveguide networks for the wave equation: stability, passivity, and numerical dispersion}, journal = {IEEE Trans. Speech and Audio Processing}, volume = {11}, number = {3}, year = {2001}, month = {may}, pages = {255{\textendash}266}, author = {Stefan Bilbao and Julius O. Smith} } @article {4717, title = {Generalized Wave Digital Filter Realizations of Arbitrary Reciprocal Connection Networks}, journal = {IEEE Transactions on Circuits and Systems I: Regular Papers}, volume = {66}, year = {2019}, note = {\ }, month = {09/2018}, pages = {694{\textendash}707}, chapter = {694}, abstract = {\ }, author = {Alberto Bernardini and Kurt James Werner and Smith III, Julius O. and Augusto Sarti} } @article {3472, title = {Instinct-Based Mating in Genetic Algorithms Applied to the Tuning of 1-NN Classifiers}, journal = {IEEE Transactions on Knowledge and Data Engineering}, volume = {22}, year = {2010}, note = {\ }, month = {12/2010}, abstract = {\ }, author = {Thiago Quirino and Miroslav Kubat and Nicholas Bryan} } @article {2569, title = {Interactive Network Media: A Dream Worth Dreaming?}, journal = {Organised Sound}, year = {2007}, pages = {10(3)209-219}, keywords = {media, music, network audio, performance}, author = {Ajay Kapur and Ge Wang and Philip P. Davidson and Cook, Perry} } @article { caceres2010:jacktripsoundwire-cmj, title = {JackTrip/SoundWIRE Meets Server Farm}, journal = {Computer Music Journal}, volume = {34}, number = {3}, year = {2010}, author = {C{\'a}ceres, Juan-Pablo and Chafe, Chris} } @article {2590, title = {The Laptop Orchestra as Classroom}, journal = {Computer Music Journal}, year = {2008}, pages = {32(1):26-37}, abstract = {

This article chronicles our pedagogical adventures in the Princeton Laptop Orchestra (PLOrk).We examine and evaluate the PLOrk classroom as well as new approaches and tools for teaching.In doing so, we present an integrated, naturally interdisciplinary teaching/learning environment for computer science, music, and performance.In such an environment, the learning and internalization of technical knowledge happens symbiotically with the acquisition of aesthetic and artistic awareness; there is only one explicit goal: learn to make compelling computer-mediated music together in an academic setting \— all other learning happens \‘along-the-way\’.The presence of experienced guest composers, who compose and teach the ensemble their new works, allows students to learn about and experiment with varying aesthetic and technical approaches while working with the composers on their new pieces. We believe this is an exciting new environment where the learning of interdisciplinary knowledge is not only natural, but also inevitable (and fun).

}, keywords = {ChucK, classroom, laptop orchestra, plork, princeton, slork}, author = {Ge Wang and Dan Trueman and Smallwood, Scott and Cook, Perry} } @article {2565, title = {Maps and Legends : Designing FPS-Based Interfaces for Multi-user Composition, Improvisation and Immersive Performance}, journal = {Lecture Notes in Computer Science - Computer Music Modeling and Retrieval. Sense of Sounds}, volume = {Volume 4969/2008}, year = {2008}, pages = {478-486}, abstract = {

This paper describes an interactive multi-channel multi-user networked system for real-time composition and immersive performance built using a modified version of the Quake III gaming engine. By tracking users\’ positional and action data within a virtual space, and by streaming that data over a network using OSC messages formatted as UDP packets to a multi-channel Pure Data patch, actions in virtual space are correlated to sonic output in a physical space. Virtual environments designed as abstract compositional maps or representative models of the users\’ actual physical space are investigated as means to guide and shape compositional and performance choices. This paper analyzes both the technological concerns for building and realizing the system as well as the compositional and perceptual issues inherent in the project itself.

}, isbn = {978-3-540-85034-2}, issn = {0302-9743 (Print) 1611-3349 (Online)}, doi = {10.1007/978-3-540-85035-9}, url = {http://www.springerlink.com/content/c2lqh551681440l7/}, author = {Rob Hamilton} } @article {4342, title = {Modal Processor Effects Inspired by Hammond Tonewheel Organs}, journal = {Applied Sciences}, year = {2016}, note = {\ }, abstract = {\ }, url = {http://www.mdpi.com/2076-3417/6/7/185}, author = {Kurt James Werner and Abel, Jonathan S} } @article {4676, title = {Modeling Circuits with Arbitrary Topologies and Active Linear Multiports using Wave Digital Filters}, journal = {IEEE Transactions on Circuits and Systems I: Regular Papers}, year = {2018}, note = {\ }, month = {06/2018}, abstract = {The wave digital filter (WDF) technique derives digital filters from analog prototypes that classically have been restricted to passive circuits with series/parallel topologies. Since most audio circuits contain active elements (e.g., op-amps) and complex topologies, WDFs have only had limited use in virtual analog modeling. In this paper, we extend the WDF approach to accommodate the unbounded class of non-series/parallel junctions which may absorb linear multiports. We present four modified-nodal-analysis-based scattering matrix derivations for these junctions, using parametric waves with voltage, power, and current waves as particular cases. Three derivations afford implementations whose cost in multiplies are lower than multiplying by the scattering matrix. Negative port resistances may be needed in WDF modeling of active circuits, restricting the WDF to voltage or current waves. We propose two techniques for localizing this restriction. Case studies on the Baxandall tone circuit and a \"Frequency Booster\" guitar pedal demonstrate the proposed techniques in action.}, keywords = {active circuits, circuit simulation, physical modeling, scattering., virtual analog, Wave digital filters}, doi = {10.1109/TCSI.2018.2837912}, url = {https://ieeexplore.ieee.org/document/8371321/}, author = {Kurt James Werner and Alberto Bernardini and Smith III, Julius O. and Augusto Sarti} } @article {4336, title = {Modeling Nonlinear Wave Digital Elements using the Lambert Function}, journal = {IEEE Transactions on Circuits and Systems I: Regular Papers}, volume = {63}, year = {2016}, note = {\ }, month = {08/2016}, chapter = {1231{\textendash}1242}, abstract = {\ }, author = {Alberto Bernardini and Kurt James Werner and Augusto Sarti and Smith III, Julius O.} } @article {2577, title = {Musical Tapestry: Re-composing Natural Sounds}, journal = {Journal of New Music Research}, year = {2008}, pages = {36(4):241-250}, abstract = {

A system to aid composition with analysis, transformation, and re-synthesis of natural sounds is described. Sinusoidal analysis is used to isolate and extract deterministic sounds, and transients are also isolated/extracted, leaving the stochastic background which is parameterized by wavelet tree analysis. All of these components become templates for the synthesis phase, which is controlled (1) by placing templates on timelines or in groups, (2) by real-time manipulation of parameters, and (3) via scripting using the ChucK language. The result is a flexible \“workbench\” for doing modern day musique concr\ète or acousmatic composition, sound design, and other sonic sculpting tasks.

}, keywords = {resynthesis, sound scene analysis, tapestrea, taps, transformation}, url = {http://taps.cs.princeton.edu/}, author = {Misra, Ananya and Ge Wang and Cook, Perry} } @article {4501, title = {Ocarina: Designing the iPhone{\textquoteright}s Magic Flute}, journal = {Computer Music Journal}, volume = {38}, year = {2014}, note = {\ }, month = {05/2014}, type = {design, engineering, social}, chapter = {8-21}, abstract = {Ocarina, created in 2008 for the iPhone, is one of the first musical artifacts in the age of pervasive, app-based mobile computing. It presents a flute-like physical interaction using microphone input, multi-touch, and accelerometers\—and a social dimension that allows users to listen in to each other around the world. This article chronicles Smule\’s Ocarina as a mobile musical experiment for the masses, examining in depth its design, aesthetics, physical interaction, and social interaction, as well as documenting its inextricable relationship with the rise of mobile computing as catalyzed by mobile devices such as the iPhone.}, keywords = {design, iphone, mobile music, musical expression, ocarina, social design}, doi = {10.1162/COMJ_a_00236}, url = {http://www.gewang.com/ocarina/}, author = {Ge Wang} } @article {2675, title = {Physical Modeling Using Digital Waveguides}, journal = {Computer Music Journal}, volume = {16}, year = {1992}, month = {12/1992}, chapter = {74}, keywords = {digital, digital waveguides, modeling, physical modeling, string, waveguides}, author = {Julius O. Smith} } @article { abelandsmithdafx06, title = {Robust Design of Very High-Order Allpass Dispersion Filters}, journal = {Proc. Conf. Digital Audio Effects (DAFx-06), Montreal, Canada}, year = {2006}, note = {http://www.dafx.de/}, month = {Sept.}, author = {Jonathan Abel and Julius O. Smith} } @article {3551, title = {SCORE-to-Humdrum: A Graphical Environment for Musicological Analysis}, journal = {Computing in Musicology}, volume = {10}, year = {1996}, note = {\ }, pages = {122}, chapter = {105}, abstract = {\ }, author = {Kornstaedt, A.} } @article { abelsmythandsmithdafx03, title = {A Simple, Accurate Wall Loss Filter for Acoustic Tubes}, journal = {Proc. Conf. Digital Audio Effects (DAFx-03), Queen Mary, University of London}, year = {2003}, note = {http://www.dafx.de/}, month = {Sept.}, author = {Jonathan Abel and Tamara Smyth and Julius O. Smith} } @article {2559, title = {The Simulation of Moving Sound Sources}, journal = {Journal of the Audio Engineering Society}, year = {1971}, pages = {J. Audio Eng. Soc. 19 (1), 2{\textendash}6.}, abstract = {A digital computer was used to generate four channels of information which are recorded on a tape recorder. The com- puter program provides control over the apparent location and movement of a synthesized sound in an illusory acoustical space. The method controls the distribution and amplitude of direct and reverberant signals between the loudspeakers to provide the angular and distance information, and introduces a Doppler shift to enhance velocity information.}, keywords = {4-channel, angular cue, distance cue, Doppler, localization, quad, spatialization, velocity cue}, author = {John Chowning} } @article {4343, title = {Sinusoidal Parameter Estimation Using Quadratic Interpolation Around Power-Scaled Magnitude Spectrum Peaks}, journal = {Applied Sciences}, year = {2016}, note = {\ }, abstract = {\ }, url = {http://www.mdpi.com/2076-3417/6/10/306}, author = {Kurt James Werner and Fran{\c c}ois G. Germain} } @article {4503, title = {Social Composition: Musical Data Systems for Expressive Mobile Music}, journal = {Leonardo Music Journal}, volume = {21}, year = {2011}, note = {\ }, month = {12/2011}, type = {scientific, aesthetics}, chapter = {57-64}, abstract = {This article explores the role of symbolic score data in the authors{\textquoteright} mobile music-making applications, as well as the social sharing and community-based content creation workflows currently in use on their online musical network. Web-based notation systems are discussed alongside in-app visual scoring methodologies for the display of pitch, timing and duration data for instrumental and vocal performance. User-generated content and community-driven ecosystems are considered alongside the role of cloud-based services for audio rendering and streaming of performance data.}, keywords = {mobile music, social composition, social music}, doi = {10.1162/LMJ_a_00062}, author = {Robert Hamilton and Jeffery Smith and Ge Wang} } @article { berdahlandsmithdafx06, title = {Some Physical Audio Effects}, journal = {Proc. Conf. Digital Audio Effects (DAFx-06), Montreal, Canada}, year = {2006}, note = {http://www.dafx.de/}, month = {Sept.}, author = {Edgar Berdahl and Julius O. Smith} } @article {4498, title = {Some Principles of Visual Design for Computer Music}, journal = {Leonardo Music Journal}, volume = {26}, year = {2016}, note = {\ }, month = {12/2016}, type = {design, aesthetics, engineering}, chapter = {14-19}, abstract = {This article presents observations pertaining to expressive visual design for computer music, focusing in particular on real-time integration of graphics and audio. The author describes specific projects as examples supporting a set of design principles that range from {\textquotedblleft}user-oriented{\textquotedblright} to {\textquotedblleft}aesthetic{\textquotedblright} and other observations. Examples include audio visualization, game-like interfaces, and mobile musical instruments.}, doi = {10.1162/LMJ_a_00960}, author = {Ge Wang} } @article {2772, title = {Spectrum Analysis Tutorial - Parts I and II}, journal = {Computer Music Journal}, volume = {11}, number = {STAN-M-33}, year = {1987}, chapter = {9}, keywords = {analysis, spectrum, tutorial}, url = {https://ccrma.stanford.edu/files/papers/stanm33.pdf}, author = {Jaffe, David A.} } @article {4702, title = {Speech Denoising with Deep Feature Losses}, journal = {arXiv:1806.10522}, year = {2018}, note = {\ }, month = {06/2018}, type = {arXiv eprint}, abstract = {

We present an end-to-end deep learning approach to denoising speech signals by processing the raw waveform directly. Given input audio containing speech corrupted by an additive background signal, the system aims to produce a processed signal that contains only the speech content. Recent approaches have shown promising results using various deep network architectures. In this paper, we propose to train a fully-convolutional context aggregation network using a deep feature loss. That loss is based on comparing the internal feature activations in a different network, trained for acoustic environment detection and domestic audio tagging. Our approach outperforms the state-of-the-art in objective speech quality metrics and in large-scale perceptual experiments with human listeners. It also outperforms an identical network trained using traditional regression losses. The advantage of the new approach is particularly pronounced for the hardest data with the most intrusive background noise, for which denoising is most needed and most challenging.
CodeAudio examples

}, url = {https://arxiv.org/abs/1806.10522}, author = {Fran{\c c}ois G. Germain and Qifeng Chen and Vladlen Koltun} } @article {3464, title = {The Switched Convolution Reverberator}, journal = {The Journal of the Audio Engineering Society}, volume = {60}, year = {2012}, note = {\ }, month = {04/2012}, pages = {227-236}, chapter = {227}, abstract = {An artificial reverberator having low memory requirements and small computational cost is presented. The reverberator consists of an equalized comb filter driving a convolution with a short noise sequence, which can consist of Gaussian or velvet noise. The reverberator\ equalization and decay rate are controlled by low-order IIR filters, and the echo density is\ that of the noise sequence. While this structure is efficient and readily generates high echo\ densities, if a fixed noise sequence is used, the reverberator has an unwanted periodicity at the\ comb filter delay length. To overcome this difficulty, the noise sequence is regularly updated\ or \“switched.\” Several structures for updating the noise sequence, including a leaky integrator\ sensitive to the signal crest factor, and a multiband architecture, are described.}, author = {Lee, Keun Sup and Abel, Jonathan S and Vesa V{\"a}lim{\"a}ki and Stilson, Timothy and David P. Berners} } @article { caceres2008:synchronization, title = {Synchronization and acoustics in network performance.}, journal = {The Journal of the Acoustical Society of America}, volume = {124}, number = {4}, year = {2008}, note = {156th Meeting of the Acoustical Society of America (ASA), Miami, Florida, November 11}, pages = {2489}, author = {C{\'a}ceres, Juan-Pablo} } @article {2790, title = {The Synthesis of Complex Audio Spectra by Means of Discrete Summation Formulae}, journal = {Journal of the Audio Engineering Society}, volume = {24}, number = {STAN-M-5}, year = {1976}, chapter = {717}, keywords = {complex audio spectra, discrete summation, formula, spectra, synthesis}, url = {https://ccrma.stanford.edu/files/papers/stanm5.pdf}, author = {Moorer, James A.} } @article {2560, title = {The Synthesis of Complex Audio Spectra by Means of Frequency Modulation}, journal = {Journal of the Audio Engineering Society}, year = {1973}, pages = {J. Audio Eng. Soc. 21 (7), 526-534.}, abstract = {A new application of the well-known process of frequency modulation is shown to result in a surprising control of audio spectra. The technique provides a means of great simplicity to control the spectral components and their evolution in time. Such dynamic spectra are diverse in their subjective impressions and include sounds both known and unknown.}, keywords = {dynamic spectra, FM synthesis, frequency modulation synthesis}, author = {John Chowning} } @article { chafe2010:musical, title = {Temporal Separation in a Rhythmic Task}, journal = {Perception}, year = {2010}, note = {Accepted for publication}, author = {Chafe, Chris and C{\'a}ceres, Juan-Pablo} } @article {3553, title = {Themefinder: A Web-based Melodic Search Tool}, journal = {Computing in Musicology }, volume = {11}, year = {1998}, note = {\ }, pages = {231-236}, abstract = {\ }, author = {Kornstaedt, A.} } @article {4499, title = {World Stage: Crowdsourcing Paradigm for Expressive Social Mobile Music}, journal = {Journal of New Music Research}, volume = {44}, year = {2015}, note = {\ }, month = {11/2014}, type = {design, engineering, mobile and social music}, chapter = {112-128}, abstract = {The combination of powerful mobile devices and the connective potential of cloud-based computing are changing how, where, and when people use computers. No longer physically tethered, computers embedded in mobile phones and tablets freely roam in daily life alongside their human users while persistently connected to the network. At the same time, the pervasiveness of computing enables us to engage people to solve large-scale problems by leveraging human intelligence and the effect of large crowds. In this paper, we discuss a new model for crowdsourced musical interactions based on mobile devices. This model, which we call World Stage, is a pseudo-anonymous, location-aware ecosystem designed to connect hundreds of thousands of users in a social-musical game involving expressive musical performance and collaborative musical feedback. We describe the motivation and mechanics of the World Stage, and present a full implementation and case study around a commercial iPhone application: Smule\’s Leaf Trombone: World Stage. We also present the experiential design and the technological infrastructure around the World Stage and discuss the unique social/musical possibilities it affords. This work is about exploring, perhaps for the very first time, a crowdsourcing eco-system that incorporates expressive music-making and with game-like elements, aimed at inciting a mass audience.}, keywords = {crowdsourcing, leaf trombone, mobile music, musical expression, social music, world stage}, doi = {DOI 10.1162/LMJ_a_00960}, author = {Ge Wang and Spencer Salazar and Jieun Oh and Robert Hamilton} } @article { caceres2010:jacktrip-jnmr, title = {{J}ack{T}rip: Under the Hood of an Engine for Network Audio}, journal = {Journal of New Music Research}, volume = {38}, number = {3}, year = {2010}, author = {C{\'a}ceres, Juan-Pablo and Chafe, Chris} } @booklet { smithfirdesign, title = {FIR Digital Filter Design}, year = {1992}, note = {Mathematica notebook for Music 420 (EE 367A), ccrma-ftp}, month = {Winter}, author = {Julius O. Smith} } @booklet { smithfauststrings, title = {Making Virtual Electric Guitars and Associated Effects Using Faust [DRAFT]}, year = {2007}, note = {http://ccrma.stanford.edu/realsimple/faust\_strings/}, month = {dec}, author = {Julius O. Smith} } @booklet { smithfaustintro07, title = {Signal Processing in Faust and Pd}, year = {2007}, note = {http://ccrma.stanford.edu/realsimple/faust/}, month = {nov}, author = {Julius O. Smith} } @booklet { berdahlandsmithtfm, title = {Transfer Function Measurement Toolbox}, year = {2007}, note = {http://ccrma.stanford.edu/realsimple/imp\_meas/}, author = {Edgar Berdahl and Julius O. Smith} } @article {2683, title = {The 2003 CCRMA Open House}, year = {2003}, keywords = {Open House, Overview, Recent Papers}, author = {Scavone, Gary P.} } @article {2778, title = {Absolute Memory for Musical Pitch: More Than The Melody Lingers On}, year = {1992}, month = {06/1992}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {melody, memory, musical pitch, pitch}, url = {https://ccrma.stanford.edu/files/papers/stanm79.pdf}, author = {Levitine, D. J.} } @article { abemoto:stanm118, title = {AM/FM Rate Estimation and Bias Correction for Time-Varying Sinusoidal Modeling}, number = {STAN-M-118}, year = {2004}, institution = {Stanford University, Department of Music}, keywords = {AM/FM, bias correction, rate estimation, sinusoidal modeling}, url = {https://ccrma.stanford.edu/files/papers/stanm118.pdf}, author = {Mototsugu Abe and Julius O. Smith} } @article {2783, title = {The Aquisition of Musical Percepts with a New Scale}, year = {1987}, month = {07/1987}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {acquisition, music, new scale, percepts}, url = {https://ccrma.stanford.edu/files/papers/stanm40.pdf}, author = {Mathews, Max Vernon and Pierce, John R.} } @article { chowning:stanm12, title = {Auditory Distance Perception Under Natural Sounding Conditions, {Final Report}}, number = {STAN-M-12}, year = {1982}, institution = {Stanford University Department of Music}, keywords = {auditory distance perception, perception}, url = {http://ccrma.stanford.edu/files/papers/stanm12.pdf}, author = {John Chowning and Christopher Sheeline} } @article {2686, title = {Automatic Species Counterpoint}, year = {1984}, month = {05/1984}, institution = {CCRMA}, address = {Stanford, CA}, keywords = {Automatic Species Counterpoint, Counterpoint, Data}, url = {https://ccrma.stanford.edu/files/papers/stanm19.pdf}, author = {Schottstaedt, Bill} } @article {2818, title = {The Bounded-Q approach to time-varying spectral analysis}, year = {1986}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {analysis, bounded-q, spectral, spectral analysis, time-varying}, url = {https://ccrma.stanford.edu/files/papers/stanm28.pdf}, author = {Mont-Reynaud, Bernard} } @article { chafe:bowed, title = {Bowed String Synthesis and Its Control From a Physical Model}, number = {STAN-M-32, replaced by STAN-M-48}, year = {1985}, note = {replaced by {STAN-M-48} {Simulating Performance on a Bowed Instrument}}, month = {07/1985}, institution = {Stanford University Department of Music}, keywords = {bowed string, physical model, synthesis}, url = {https://ccrma.stanford.edu/files/papers/stanm48.pdf}, author = {Chafe, Chris} } @article {2687, title = {CCRMA Overview April 2003}, year = {2003}, month = {04/2003}, institution = {CCRMA}, address = {Stanford, CA}, keywords = {Overview}, issn = {STAN-M-112}, url = {https://ccrma.stanford.edu/files/papers/stanm112.pdf}, author = {Scavone, Gary P.} } @article {2803, title = {CCRMA Overview, April 2000}, year = {2000}, month = {04/2000}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, url = {https://ccrma.stanford.edu/files/papers/stanm106.pdf}, author = {Leistikow, R. and Scavone, Gary P.} } @article {2811, title = {CCRMA Overview, April 2002}, year = {2002}, month = {04/2002}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, author = {Scavone, Gary P.} } @article {2810, title = {CCRMA Overview, February 1999}, year = {1999}, month = {02/1999}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, url = {http://ccrma.stanford.edu/files/papers/stanm105.pdf}, author = {Scavone, Gary P.} } @article {2809, title = {CCRMA Overview, January 1998}, year = {1998}, month = {01/1998}, institution = {Stanford University}, address = {Stanford, CA}, keywords = {Overview}, author = {Scavone, Gary P.} } @article {2805, title = {CCRMA Overview, July 2001}, year = {2001}, month = {07/2001}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, url = {https://ccrma.stanford.edu/files/papers/stanm107.pdf}, author = {Nichols, C.} } @article {2807, title = {CCRMA Publications, 1970-1995}, year = {1996}, month = {01/1996}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, url = {http://ccrma.stanford.edu/files/papers/stanm103.pdf}, author = {Scavone, Gary P.} } @article {2802, title = {CCRMA Report, May 1996}, year = {1996}, month = {05/1996}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, url = {https://ccrma.stanford.edu/files/papers/stanm98.pdf}, author = {Igoudin, Alex and Julius O. Smith} } @article { chowning:stanm1, title = {Computer Simulation of Music Instrument Tones in Reverberant Environments}, number = {STAN-M-1}, year = {1974}, institution = {Stanford University Department of Music}, keywords = {computer simulation, music instrument tones, reverberant environments}, url = {http://ccrma.stanford.edu/files/papers/stanm1.pdf}, author = {John Chowning and Grey, John M. and Loren Rush and Moorer, James A.} } @article {2782, title = {The Conductor Program and Mechanical Baton}, year = {1988}, month = {05/1988}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {baton, conductor, mechanical baton}, url = {https://ccrma.stanford.edu/files/papers/stanm47.pdf}, author = {Mathews, Max Vernon and Barr, D.} } @article { chaigne:stanm119, title = {Coupled Structural-Acoustic Systems. Application to Stringed Instruments}, number = {STAN-M-119}, year = {2004}, institution = {Stanford University, Department of Music}, keywords = {coupled structural-acoustic systems, stringed instruments}, url = {https://ccrma.stanford.edu/files/papers/stanm119.pdf}, author = {Antoine Chaigne} } @article { abemoto:stanm117, title = {CQIFFT: Correcting Bias in a Sinusoidal Parameter Estimator based on Quadratic Interpolation of FFT Magnitude Peaks}, number = {STAN-M-117}, year = {2004}, institution = {Stanford University, Department of Music}, keywords = {FFT magnitude peaks, sinusoidal parameter estimator}, url = {http://ccrma.stanford.edu/files/papers/stanm117.pdf}, author = {Mototsugu Abe and Julius O. Smith} } @article { abemoto:stanm114, title = {Design Criteria for the Quadratically Interpolated FFT Method (I): Bias due to Interpolation}, number = {STAN-M-114}, year = {2004}, institution = {Stanford University, Department of Music}, keywords = {FFT method, interpolation, quadratically interpolated}, url = {https://ccrma.stanford.edu/files/papers/stanm114.pdf}, author = {Mototsugu Abe and Julius O. Smith} } @article { abemoto:stanm115, title = {Design Criteria for the Quadratically Interpolated FFT Method (II): Bias due to Interfering Components}, number = {STAN-M-115}, year = {2004}, institution = {Stanford University, Department of Music}, keywords = {FFT method, interfering components, quadratically interpolated}, url = {https://ccrma.stanford.edu/files/papers/stanm115.pdf}, author = {Mototsugu Abe and Julius O. Smith} } @article { abemoto:stanm116, title = {Design Criteria for the Quadratically Interpolated FFT Method (III): Bias due to Amplitude and Frequency Modulation}, number = {STAN-M-116}, year = {2004}, institution = {Stanford University, Department of Music}, keywords = {amplitude, FFT method, frequency modulation, quadratically interpolated}, url = {https://ccrma.stanford.edu/files/papers/stanm116.pdf}, author = {Mototsugu Abe and Julius O. Smith} } @article {2758, title = {Designing Digital Instruments with Hierarchical Waveguide Networks}, year = {1988}, month = {05/1988}, institution = {Stanford University}, address = {Stanford, California}, keywords = {design, digital instruments, waveguide networks}, url = {https://ccrma.stanford.edu/files/papers/stanm49.pdf}, author = {Diener, G.} } @article {2797, title = {An Environment for the Analysis, Transformation and Resynthesis of Musical Sounds}, year = {1988}, month = {05/1988}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {analysis, resynthesis, transformation}, url = {https://ccrma.stanford.edu/files/papers/stanm52.pdf}, author = {Serra, Xavier} } @article {2647, title = {Feedback Control of Acoustic Musical Instruments}, number = {120}, year = {2008}, month = {06/2008}, institution = {Stanford University}, address = {Stanford, CA}, url = {https://ccrma.stanford.edu/~eberdahl/Papers/STANM120.pdf}, author = {Edgar Berdahl and Gunter Niemeyer and Julius O. Smith} } @article { carter:scores, title = {A Generalized Approach to Automatic Recognition of Music Scores}, number = {STAN-M-87}, year = {1993}, month = {12/1993}, institution = {Stanford University Department of Music}, address = {Stanford, California}, keywords = {automatic recognition, music scores, recognition}, url = {https://ccrma.stanford.edu/files/papers/stanm87.pdf}, author = {Nicholas Carter} } @article {2792, title = {High Level Factors and the Musical Saliency of Auditory Phenomena}, year = {1987}, month = {07/1987}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {auditory phenomena, musical saliency}, url = {https://ccrma.stanford.edu/files/papers/stanm41.pdf}, author = {Pierce, John R. and Schubert, E. D.} } @article {2774, title = {History and Principles of Microtonal Keyboard Design}, year = {1988}, month = {04/1988}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {design, history, keyboard, microtonal, microtonal keyboard}, url = {https://ccrma.stanford.edu/files/papers/stanm45.pdf}, author = {Keislar, D.} } @article { cook:bore, title = {Implementation of Single Reed Instruments with Arbitrary Bore Shapes Using Digital Waveguide Filters}, number = {STAN-M-50}, year = {1988}, month = {05/1988}, institution = {Stanford University Department of Music}, keywords = {digital waveguide filters, single reed instruments}, url = {http://ccrma.stanford.edu/files/papers/stanm50.pdf}, author = {Cook, Perry} } @article { chowning:stanm11, title = {Instrumental Timbre and Related Acoustical Phenomena in the Perception of Music, Final Report}, number = {STAN-M-11}, year = {1982}, month = {08/1982}, institution = {Stanford University Department of Music}, address = {Stanford, California}, keywords = {acoustics, instrumental timbre, perception, timbre}, url = {http://ccrma.stanford.edu/files/papers/stanm11.pdf}, author = {John Chowning and Grey, John M. and Moorer, James A. and Loren Rush} } @article { chowning:stanm36, title = {Intelligent Analysis of Composite Acoustic Signals, Final Report}, year = {1986}, month = {05/1986}, institution = {Stanford University Department of Music}, address = {Stanford, California}, keywords = {acoustic signals, composite}, url = {http://ccrma.stanford.edu/files/papers/stanm36.pdf}, author = {John Chowning and Mont-Reynaud, Bernard} } @article { chowning:stanm15, title = {Intelligent Systems for the Analysis of Digitized Acoustic Signals, Final Report}, number = {STAN-M-15}, year = {1984}, institution = {Stanford University Department of Music}, keywords = {analysis, digital acoustic signals}, url = {http://ccrma.stanford.edu/files/papers/stanm15.pdf}, author = {John Chowning and Loren Rush and Mont-Reynaud, Bernard and Chafe, Chris and Schloss, W. A. and Julius O. Smith} } @article {2691, title = {Learning About Harmony with Harmony Space: an Overview}, year = {1994}, month = {09/1994}, institution = {CCRMA}, address = {Stanford, CA}, keywords = {composing, Harmony Space}, issn = {STAN-M-88}, url = {https://ccrma.stanford.edu/files/papers/stanm88.pdf}, author = {Holland, Simon} } @article {2764, title = {The Moose: A Haptic User Interface for Blind Persons with Application to the Digital Sound Studio}, year = {1995}, month = {10/1995}, institution = {Stanford University}, address = {Stanford, California}, keywords = {blind, digital sound, haptic, moose, studio, user interface}, url = {https://ccrma.stanford.edu/files/papers/stanm95.pdf}, author = {Gillespie, Brent and O{\textquoteright}Modhrain, Sile} } @article {2751, title = {Music Applications of Digital Waveguides}, year = {1987}, institution = {Stanford University}, keywords = {Applications, digital, music, waveguides}, url = {https://ccrma.stanford.edu/files/papers/stanm39.pdf}, author = {Julius O. Smith} } @article { chowning:stanm64, title = {Music From Machines: Perceptual Fusion and Auditory Perspective - for Ligeti}, number = {STAN-M-64}, year = {1990}, institution = {Stanford University Department of Music}, keywords = {auditory perspective, ligetic, machines, music, perceptual fusion}, url = {http://ccrma.stanford.edu/files/papers/stanm64.pdf}, author = {John Chowning} } @article {2801, title = {On Fast FIR Filters Implemented as Tail-Canceling IIR Filters}, year = {1994}, month = {11/1994}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Fasr FIR Filters, IIR Filters, Tail-Cancelling}, url = {https://ccrma.stanford.edu/files/papers/stanm90.pdf}, author = {Wang, Avery Li-Chun and Julius O. Smith} } @article {2787, title = {On the Loudness of Complex, Time-Variant Tones}, year = {1975}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {complex, loudness, time-variant, tones}, url = {https://ccrma.stanford.edu/files/papers/stanm4.pdf}, author = {Moorer, James A.} } @article {2815, title = {Overview, September 1994}, year = {1994}, month = {09/1994}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, author = {Van Duyne, Scott A. and Wood, Patte} } @article {2798, title = {OVERVIEW: Center for Computer Research in Music and Acoustics (Recent Work)}, year = {1988}, month = {03/1988}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {Overview}, url = {https://ccrma.stanford.edu/files/papers/stanm44.pdf}, author = {Serra, Xavier and Wood, Patte} } @article {2781, title = {Pickups for the Vibrations of Violin and Guitar Strings Using Piezoelectric Bimorphic Bender Elements}, year = {1988}, month = {05/1988}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {bender, bimorphic, guitar, pickups, piezoelectric, Strings, vibrations, Violin}, url = {https://ccrma.stanford.edu/files/papers/stanm54.pdf}, author = {Mathews, Max Vernon} } @article {2812, title = {PLA - A Tutorial and Reference Manual}, year = {1984}, month = {12/1984}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {manual, PLA, tutorial}, url = {https://ccrma.stanford.edu/files/papers/stanm24.pdf}, author = {Schottstaedt, Bill} } @article { chafe:stanm65, title = {Pulsed Noise and Microtransients in Physical Models of Musical Instruments}, number = {STAN-M-65}, year = {1990}, month = {sep}, institution = {Stanford University Department of Music}, keywords = {microtransients, musical instruments, physical models, pulsed noise}, url = {https://ccrma.stanford.edu/files/papers/stanm65.pdf}, author = {Chafe, Chris} } @article { bosi:spatial, title = {A Real-Time System for Spatial Distribution of Sound}, number = {STAN-M-66}, year = {1990}, month = {08/1990}, institution = {Stanford University Department of Music}, keywords = {real-time, spatial, spatial distribution}, url = {https://ccrma.stanford.edu/files/papers/stanm66.pdf}, author = {Marina Bosi} } @article {2776, title = {Recognition of Complex Auditory-Spatial Patterns}, year = {1991}, month = {06/1991}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {auditory-spatial paterns}, url = {https://ccrma.stanford.edu/files/papers/stanm71.pdf}, author = {Lakatos, S.} } @article {2735, title = {Reverberation Cancellation in Musical Signals Using Adaptive Filters}, number = {STAN-M-51}, year = {1988}, month = {05/1988}, institution = {CCRMA}, address = {Stanford, California}, keywords = {Adaptive Filters, filters, reverberation, Reverberation Cancellation}, url = {https://ccrma.stanford.edu/files/papers/stanm51.pdf}, author = {Cook, Perry} } @article {2674, title = {The Role of Psychoacoustics in Computer Music}, year = {1987}, month = {06/1987}, institution = {CCRMA}, address = {Stanford, CA}, keywords = {computer, computer music, music, psychoacoustics}, url = {https://ccrma.stanford.edu/files/papers/stanm38.pdf}, author = {John W. Gordon} } @article { chowning:stanm8, title = {Simulation of Music Instrument Tones in Reverberant Environments, Final Report}, year = {1978}, month = {05/1978}, institution = {Stanford University Department of Music}, keywords = {musical instruments, reverberant environments, simulation}, issn = {STAN-M-8}, url = {http://ccrma.stanford.edu/files/papers/stanm8.pdf}, author = {John Chowning and Grey, John M. and Loren Rush and Moorer, James A. and Leland Smith} } @article {2777, title = {Temporal Constraints on Apparent Motion in Auditory Space}, year = {1991}, month = {11/1991}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {apparent motion, constraints, psychoacoustics, space, temporal constraints}, url = {https://ccrma.stanford.edu/files/papers/stanm74.pdf}, author = {Lakatos, S.} } @article {2806, title = {Towards a Better Software-Design for Supporting Creative Musical Activity (CMA)}, year = {1993}, month = {10/1993}, institution = {Stanford University, CCRMA}, address = {Stanford, CA}, keywords = {CMA, creative musical activity, software-design}, url = {https://ccrma.stanford.edu/files/papers/stanm84.pdf}, author = {Oppenheim, Daniel V.} } @article {2759, title = {TTREES: An Active Data Structure for Computer Music}, year = {1988}, month = {05/1988}, institution = {Stanford University}, address = {Stanford, California}, keywords = {computer music, data structure, music, TTREES}, url = {https://ccrma.stanford.edu/files/papers/stanm53.pdf}, author = {Diener, G.} } @article {5883, title = {DawDreamer: Bridging the Gap Between Digital Audio Workstations and Python Interfaces}, year = {2021}, note = {\ }, abstract = {Audio production techniques which previously only existed in GUI-constrained digital audio workstations, livecoding environments, or C++ APIs are now accessible with our new Python module called DawDreamer. DawDreamer therefore bridges the gap between real sound engineers and coders imitating them with offline batch-processing. Like contemporary modules in this domain, DawDreamer can create directed acyclic graphs of audio processors such as VSTs which generate or manipulate audio streams. DawDreamer can also dynamically compile and execute code from Faust, a powerful signal processing language which can be deployed to many platforms and microcontrollers. We discuss DawDreamer{\textquoteright}s unique features in detail and potential applications across music information retrieval including source separation, transcription, and audio effect parameter inference. We provide fully cross-platform PyPI installers, a Linux Dockerfile, and an example Jupyter notebook.}, url = {https://arxiv.org/abs/2111.09931}, author = {Braun, David} } @mastersthesis {2808, title = {An Acoustic Analysis of Single-Reed Woodwind Instruments with an Emphasis on Design}, volume = {Ph.D.}, number = {STAN-M-100}, year = {1997}, month = {03/1997}, school = {Stanford University}, address = {Stanford, CA}, keywords = {design, Digital waveguide modeling, instruments, modeling, performance, single-reed woodwind, woodwind}, url = {https://ccrma.stanford.edu/files/papers/stanm100.pdf}, author = {Scavone, Gary P.} } @mastersthesis {2589, title = {The ChucK Audio Programming Language}, volume = {PhD}, year = {2008}, school = {Princeton University}, abstract = {

This thesis argues for the philosophy, design, and development of ChucK, a general-purpose programming language tailored for computer music. The goal is to create a language that is expressive and easy to write and read with respect to time and parallelism, and to provide a platform for precise audio synthesis/analysis and rapid experimentation in computer music. In particular, ChucK provides a syntax for representing information flow, a new time-based concurrent programming model that allows programmers to flexibly and precisely control the flow of time in code (we call this \"strongly-timed\"), and facilities to develop programs on-the-fly - as they run. A ChucKian approach to live coding as a new musical performance paradigm is also described. In turn, this motivates the Audicle, a specialized graphical environment designed to facilitate on-the-fly programming, to visualize and monitor ChucK programs in real-time, and to provide a platform for building highly customizable user interfaces.

}, keywords = {audio, ChucK, ge wang, phd thesis, programming language}, url = {http://www.cs.princeton.edu/~gewang/thesis.html}, author = {Ge Wang} } @mastersthesis {2768, title = {Digital Waveguide Modeling and Simulation of Reed Woodwind Instruments}, volume = {Ph.D.}, number = {STAN-M-72}, year = {1991}, month = {07/1991}, school = {Stanford University}, address = {Stanford, CA}, keywords = {Digital waveguide modeling, instruments, modeling, simulation, woodwind}, url = {https://ccrma.stanford.edu/files/papers/stanm72.pdf}, author = {Hirschman, Suzanne} } @mastersthesis { chomyszynphd, title = {Distance of Sound in Reverberant Fields}, volume = {Ph.D.}, number = {STAN-M-94}, year = {1995}, month = {08/1995}, school = {Department of Music, Stanford University}, type = {phd}, keywords = {distance of sound, reverberant fields}, url = {https://ccrma.stanford.edu/files/papers/stanm94.pdf}, author = {Jan Chomyszyn} } @mastersthesis { borisht, title = {Electronic Simulation of Auditorium Acoustics}, volume = {Ph.D.}, number = {STAN-M-18}, year = {1984}, school = {Elec.~Eng. Dept., Stanford University}, type = {phd}, address = {Stanford, California}, keywords = {acoustics, auditorium, auditorium acoustics, electronic simulation}, url = {https://ccrma.stanford.edu/files/papers/stanm18.pdf}, author = {Jeffrey Borish} } @mastersthesis {2804, title = {Embedding Equal Pitch Spaces and The Question of Expanded Chromas: An Experimental Approach}, volume = {Ph.D.}, number = {STAN-M-93}, year = {1995}, month = {12/1995}, school = {Stanford University}, address = {Stanford, CA}, keywords = {embedding, equal pitch spaces, expanded chromas}, url = {https://ccrma.stanford.edu/files/papers/stanm93.pdf}, author = {Enrique Moreno} } @mastersthesis {2785, title = {Event Formation and Separation in Musical Sound}, volume = {Ph.D.}, number = {STAN-M-77}, year = {1991}, month = {12/1991}, school = {Stanford University}, address = {Stanford, CA}, keywords = {event formation, music, separation, sound}, url = {https://ccrma.stanford.edu/files/papers/stanm77.pdf}, author = {Mellinger, David K.} } @mastersthesis {2767, title = {An Exploration of Musical Timbre}, volume = {Ph.D.}, number = {STAN-M-2}, year = {1975}, school = {Stanford University}, address = {Stanford, California}, keywords = {musical timbre, timbre}, url = {https://ccrma.stanford.edu/files/papers/stanm2.pdf}, author = {Grey, John M.} } @mastersthesis {2763, title = {Haptic Displays of Systems with Changing Kinematic Constraints: The Virtual Piano Action}, volume = {Ph.D.}, number = {STAN-M-92}, year = {1996}, month = {01/1996}, school = {Stanford University}, address = {Stanford, California}, keywords = {displays, haptic, kinematic constraints, virtual piano action}, url = {https://ccrma.stanford.edu/files/papers/stanm92.pdf}, author = {Gillespie, Brent} } @mastersthesis {2746, title = {Identification of Control Parameters in an Articulatory Vocal Tract Model, with Applications to the Synthesis of Singing}, volume = {Ph.D.}, number = {STAN-M-68}, year = {1990}, month = {12/1990}, school = {Stanford University}, address = {Stanford, California}, keywords = {control parameters, model, synthesis of singing, vocal tract}, url = {https://ccrma.stanford.edu/files/papers/stanm68.pdf}, author = {Cook, Perry} } @mastersthesis {2770, title = {Impact of MIDI on Electroacoustic Art Music}, volume = {Ph.D.}, number = {STAN-M-102}, year = {1997}, month = {06/1997}, school = {Stanford University}, address = {Stanford, CA}, keywords = {art music, electroacoustic, MIDI}, url = {https://ccrma.stanford.edu/files/papers/stanm102.pdf}, author = {Igoudin, Alex} } @mastersthesis { cohen:thesis, title = {The Influence of Nonharmonic Partials on Tone Perception}, volume = {Ph.D.}, number = {STAN-M-10}, year = {1980}, note = {Out of print.}, school = {Stanford University}, type = {phd}, keywords = {nonharmonic partials, tone perception}, author = {Cohen, Elizabeth A.} } @mastersthesis {2814, title = {Instantaneous and Frequency-Warped Signal Processing Techniques for Auditory Source Separation}, volume = {Ph.D.}, number = {STAN-M-86}, year = {1994}, school = {Stanford University}, address = {Stanford, CA}, keywords = {auditory, signal processing, source separation}, author = {Wang, Avery Li-Chun} } @mastersthesis {3754, title = {Interactive Sound Source Separation}, volume = {PhD}, year = {2014}, note = {\ }, month = {03/2014}, pages = {165}, school = {Stanford University}, address = {Stanford}, abstract = {\ }, author = {Nicholas Bryan} } @mastersthesis {2749, title = {An Investigation of the Effects of Direct and Reverberant Signal Interactions on Auditory Distance Perception}, volume = {Ph.D.}, number = {STAN-M-13}, year = {1982}, school = {Stanford University}, type = {Ph.D.}, keywords = {auditory, Direct, Distance, Effects, Interactions, Investigation, perception, Reverberant, Signal}, url = {https://ccrma.stanford.edu/files/papers/stanm13.pdf}, author = {Christopher Sheeline} } @mastersthesis {2761, title = {Modeling Music Notation: A Three-Dimensional Approach}, volume = {Ph.D.}, number = {STAN-M-69}, year = {1990}, month = {12/1990}, school = {Stanford University}, address = {Stanford, California}, keywords = {modeling music notation, notation, three-dimensional}, url = {https://ccrma.stanford.edu/files/papers/stanm69.pdf}, author = {Diener, G.} } @mastersthesis {2753, title = {Modeling Musical Transitions}, volume = {Ph.D.}, number = {STAN-M-26}, year = {1985}, month = {06/1985}, school = {Stanford University}, type = {Ph.D.}, keywords = {modeling, Musical, Transitions}, url = {https://ccrma.stanford.edu/files/papers/stanm26.pdf}, author = {Strawn, J.} } @mastersthesis {3198, title = {A Non-negative Framework for Joint Modeling of Spectral Structure and Temporal Dynamics in Sound Mixtures}, volume = {PhD}, year = {2010}, month = {06/2010}, school = {Stanford University}, url = {https://ccrma.stanford.edu/~gautham/Site/Publications_files/gjm_thesis.pdf}, author = {Mysore, Gautham J.} } @mastersthesis {2762, title = {An Object-Oriented Real-Time Simulation of Music Performance Using Interactive Control}, volume = {Ph.D.}, number = {STAN-M-78}, year = {1991}, school = {California Institute of Technology}, keywords = {interactive control, music performance, object-oriented, real-time, simulation}, url = {https://ccrma.stanford.edu/files/papers/stanm78.pdf}, author = {Dyer} } @mastersthesis {2793, title = {On the Automatic Transcription of Percussive Music - From Acoustic Signal to High-Level Analysis}, number = {STAN-M-27}, year = {1985}, month = {05/1985}, school = {Stanford University}, address = {Stanford, CA}, keywords = {acoustic signal, analysis, automatic transciption, percussion, transcription}, url = {https://ccrma.stanford.edu/files/papers/stanm27.pdf}, author = {Schloss, W. A.} } @mastersthesis {2789, title = {On the Segmentation and Analysis of Continuous Musical Sound by Digital Computer}, volume = {Ph.D.}, number = {STAN-M-3}, year = {1975}, school = {Stanford University}, address = {Stanford, CA}, keywords = {analysis, continuous musical sound, digital computer, segmentation}, url = {https://ccrma.stanford.edu/files/papers/stanm3.pdf}, author = {Moorer, James A.} } @mastersthesis {2765, title = {Perception of Attack Transients in Musical Tones}, volume = {Ph.D.}, number = {STAN-M-17}, year = {1984}, school = {Stanford University}, address = {Stanford, California}, keywords = {attack, musical tones, perception, transients}, url = {https://ccrma.stanford.edu/files/papers/stanm17.pdf}, author = {John W. Gordon} } @mastersthesis {2775, title = {Psychoacoustic Factors in Musical Intonation: Beats, Interval Tuning, and Inharmonicity}, volume = {Ph.D.}, number = {STAN-M-70}, year = {1991}, school = {Stanford University}, address = {Stanford, CA}, keywords = {inharmonicity, intonation, musical intonation, psychoacoustics, tuning}, url = {https://ccrma.stanford.edu/files/papers/stanm70.pdf}, author = {Keislar, D.} } @mastersthesis {2786, title = {Real Time Interactive Computer Music Synthesis}, volume = {Ph.D.}, year = {1977}, school = {Stanford University}, address = {Stanford, CA}, keywords = {computer music, interactive, real time, synthesis}, url = {https://ccrma.stanford.edu/files/papers/stanm7.pdf}, author = {Moore, F. R.} } @mastersthesis {2754, title = {Some Aspects of Applying Psychoacoustic Principles to Soundfield Reconstruction}, volume = {Ph.D.}, number = {STAN-M-96}, year = {1995}, month = {12/1995}, school = {Stanford University}, type = {Ph.D.}, keywords = {Applying, Aspects, Principles, Psychoacoustic, Reconstruction, Soundfield}, url = {https://ccrma.stanford.edu/files/papers/stanm96.pdf}, author = {Trautmann, Steven D.} } @mastersthesis {2731, title = {Spectral Fusion, Spectral Parsing and the Formation of Auditory Images}, volume = {Ph.D.}, year = {1984}, month = {05/1984}, school = {Stanford University}, address = {Stanford, California}, keywords = {auditory, auditory images, hearing, spectral fusion, spectral parsing}, url = {https://ccrma.stanford.edu/files/papers/stanm22.pdf}, author = {S. McAdams} } @mastersthesis {2796, title = {A System for Sound Analysis/Transformation/Synthesis Based on a Deterministic Plus Stochastic Decomposition}, volume = {Ph.D.}, number = {STAN-M-58}, year = {1989}, school = {Stanford University}, address = {Stanford, CA}, keywords = {decomposition, determinisitic, sound analysis, stochastic, synthesis, transformation}, url = {https://ccrma.stanford.edu/files/papers/stanm58.pdf}, author = {Serra, Xavier} } @mastersthesis {2748, title = {Techniques for Digital Filter Design and System Identification with Application to the Violin}, volume = {Ph.D.}, number = {STAN-M-14}, year = {1983}, month = {06/1983}, school = {Stanford University}, keywords = {Application, design, digital, Filter, Identification, System, Techniques, Violin}, url = {https://ccrma.stanford.edu/files/papers/stanm14.pdf}, author = {Julius O. Smith} } @mastersthesis {2780, title = {Toward a High-Quailty Singing Synthesizer with Vocal Texture Control}, volume = {Ph.D.}, number = {STAN-M-111}, year = {2002}, month = {07/2002}, school = {Stanford University}, address = {Stanford, CA}, keywords = {singing, synthesizer, texture, vocal}, author = {Lu, H.-L.} } @mastersthesis {2779, title = {Towards a Theory of Timbre}, volume = {Ph.D.}, number = {STAN-M-42}, year = {1987}, month = {08/1987}, school = {Stanford University}, address = {Stanford, CA}, keywords = {timbre}, url = {https://ccrma.stanford.edu/files/papers/stanm42.pdf}, author = {Lo, Yee On} } @mastersthesis {4384, title = {Virtual Analog Modeling of Audio Circuitry Using Wave Digital Filters}, volume = {Ph.D. in Computer-Based Music Theory and Acoustics}, year = {2016}, note = {\ }, month = {12/2016}, pages = {261}, school = {Stanford University}, type = {Dissertation}, address = {Stanford}, abstract = {

The Wave Digital Filter (WDF) approach to discretizing electronic circuits has generated substantial interest in the Virtual Analog research community, which is attracted to the potential for systematic and modular simulation of vintage electronic musical instruments (e.g., synthesizers, drum machines) and audio effects (e.g., guitar amplifiers, distortion pedals). Unfortunately, traditional WDF techniques falter at the complexity of typical audio circuits, which contain complicated (non-series/parallel) topologies and multiple nonlinear devices (e.g., diodes, transistors, triodes). In this dissertation, I review classical WDF techniques and propose novel ways of systematically modeling circuits with complicated topologies and multiple nonlinearities. Throughout, the classic Bass Drum circuit from the Roland TR-808 serves as a motivating example and each of the four chapters contains a case study simulating a different one of its subcircuits.

The first two chapters consider linear reference circuits. After deriving the standard WDF building blocks and demonstrating their use in a simulation of the TR-808 Bass Drum\’s output filter, I explain their substantial limitations in the context of modeling analog audio circuitry. To overcome these limitations, I introduce a novel Modified-Nodal-Analysis-based approach to creating WDFs from circuits with complicated (non-series/parallel) topologies and show it in action on several versions of the Bass Drum\’s central circuit (the Bridged-T Resonator).

The next two chapters consider nonlinear reference circuits. I review the standard WDF technique for handling reference circuits with a single one-port nonlinear device, exemplified by a model of the TR-808 Bass Drum\’s Pulse Shaper subcircuit. In order to overcome the limitation to a single nonlinearity, I propose a novel approach that accommodates multiple nonlinearities and apply it towards a simulation of the full nonlinear version of the Bass Drum\’s Bridged-T Resonator.

Together, these four case studies form a complete model of the TR-808 Bass Drum that almost exactly matches the behavior of the real circuit, demonstrating the utility of the classical WDF techniques combined with my proposed methods. Beyond this case study, the proposed methods enable the simulation of a huge class of audio circuitry that was previously beyond the scope of WDF modeling.

}, author = {Kurt James Werner} } @mastersthesis { bilbao:thesis, title = {Wave and Scattering Methods for the Numerical Integration of Partial Differential Equation}, volume = {Ph.D.}, number = {STAN-M-108}, year = {2001}, month = {05/2001}, school = {Department of Electrical Engineering, Stanford University}, type = {phd}, address = {Stanford, California}, keywords = {numerical integration, partial differential equation, scattering, wave, wave and scattering methods}, url = {https://ccrma.stanford.edu/files/papers/stanm108.pdf}, author = {Stefan Bilbao} } @article {4120, title = {The Roland TR-808 and the Tale of the Marching Anteaters}, journal = {Sounding Board}, year = {2015}, note = {\ }, publisher = {Ethnomusicology Review}, abstract = {\ }, url = {http://ethnomusicologyreview.ucla.edu/content/roland-tr-808-and-tale-marching-anteaters}, author = {Kurt James Werner} }