Jonathan Abel
Stanford University, CCRMA / Music Department, Faculty Member
Virtual analog effects are a consequence of the ongoing digitization of all equipment used in music production. Various digital methods to imitate the warm or lo-fi sound qualities that remind listeners of analog times are covered in this... more
Virtual analog effects are a consequence of the ongoing digitization of all equipment used in music production. Various digital methods to imitate the warm or lo-fi sound qualities that remind listeners of analog times are covered in this chapter. In particular, many algorithms presented in this chapter are physical models of audio effect boxes that have been traditionally analog electronic or electromechanical devices, such as voltage-controlled filters and spring reverberation units. Some algorithms, for instance the telephone sound ...
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A signal processing method to impart the response of an acoustic string instrument to an electric instrument that includes frequency-dependent string decay alterations is proposed. This type of processing is relevant when trying to make a... more
A signal processing method to impart the response of an acoustic string instrument to an electric instrument that includes frequency-dependent string decay alterations is proposed. This type of processing is relevant when trying to make a less resonant instrument, such as an electric guitar, sound similar to a more resonant instrument, such as acoustic guitar. Unlike previous methods which typically only perform equalization, our method includes detailed physics-based string damping changes by using a time-varying filter which adds frequency-dependent exponential damping. Efficient digital filters are fit to bridge admittance measurements of an acoustic instrument and used to create equalization filters as well as damping correction filters. The damping correction filters are designed to work in real-time as they are triggered by onset and pitch detection of the signal measured through an under-saddle pickup to determine the intensity of the damping. A test case is presented in which an electric guitar is processed to model a measured acoustic guitar.
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A general, explicit, and novel formulation for Wave Digital Filters (WDFs) with multiple/multiport nonlinearities is presented. It confronts graph-theoretic views of WDF structures (SPQR tree) with techniques from the nonlinear loop... more
A general, explicit, and novel formulation for Wave Digital Filters (WDFs) with multiple/multiport nonlinearities is presented. It confronts graph-theoretic views of WDF structures (SPQR tree) with techniques from the nonlinear loop resolution literature (the K method) and a novel method for deriving scattering matrices (WDF adaptors) of arbitrarily complex topologies. It accommodates any number of memoryless nonlinearities with any number of ports each without requiring simplifying assumptions. A case study on the first clipping stage of the Big Muff Pi distortion pedal is presented. This circuit, with its multiple nonlinearities, multiport nonlinearity, and complicated topology, poses an intractable problem for state-of-the-art WDF methods. Hence, successful simulation of this circuit demonstrates the novelty and robustness of our framework.
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A recording technique due to Walter Murch for extending the reverberation time of a room is analyzed, and a realtime implementation is presented. The technique involves speeding up a prerecorded dry sound and playing it into a room. The... more
A recording technique due to Walter Murch for extending the reverberation time of a room is analyzed, and a realtime implementation is presented. The technique involves speeding up a prerecorded dry sound and playing it into a room. The room response is recorded and subsequently slowed down such that the original signal appears at its normal speed, while the reverberation of the room is ‘stretched,’ causing the room to sound larger than it is. A signal analysis is presented showing that this process is equivalent to slowing down the impulse response of the room. Measurements on a simple physical system confirm this effect, and show that the process can be interpreted as either scaling the room dimensions, or slowing the speed of sound. Finally, we describe a block processing approach which implements this technique in real time with a fixed processing latency.
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We present a Modified-Nodal-Analysis-derived method for developing Wave Digital Filter (WDF) adaptors corresponding to complicated (non-series/parallel) topologies that may include multiport linear elements (e.g. controlled sources and... more
We present a Modified-Nodal-Analysis-derived method for developing Wave Digital Filter (WDF) adaptors corresponding to complicated (non-series/parallel) topologies that may include multiport linear elements (e.g. controlled sources and transformers). A second method resolves noncomputable (non-tree-like) arrangements of series/parallel adaptors. As with the familiar 3-port series and parallel adaptors, one port of each derived adaptor may be rendered reflection-free, making it acceptable for inclusion in a standard WDF tree. With these techniques, the class of acceptable reference circuits for WDF modeling is greatly expanded. This is demonstrated by case studies on circuits which were previously intractable with WDF methods: the Bassman tone stack and Tube Screamer tone/volume stage.
Research Interests: Engineering, Music Technology, Physical modeling, Ordinary Differential Equations (ODE), Virtual Analog, and 9 moreDiscretization, Circuit simulation, Music Technology /Audio Engineering, Network Topology, Digital Audio Effects, Design of Digital filters, Traveling wave methods, Circuit Modeling, and Wave Digital Filters
Ribbon microphones are known for their warm sonics, owing in part to the unique ribbon motion induced by the sound field. Here the motion of the corrugated ribbon element in a sound field is considered, and a physical model of the ribbon... more
Ribbon microphones are known for their warm sonics, owing in part to the unique ribbon motion induced by the sound field. Here the motion of the corrugated ribbon element in a sound field is considered, and a physical model of the ribbon motion is presented. The model separately computes propagating torsional disturbances and coupled transverse and longitudinal disturbances. Each propagation mode is implemented as a mass-spring model where a mass is identified with a ribbon corrugation fold. The model is parameterized using ribbon material and geometric properties. Laser vibrometer measurements are presented, revealing stiffness in the transverse and longitudinal propagation, and showing close agreement between measured and modeled ribbon motion.
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The problem of providing an audible separation of mixed sound sources is important for a number of applications, including speech recognition, noise reduction in communication channels, re-mixing of recorded music, and using environmental... more
The problem of providing an audible separation of mixed sound sources is important for a number of applications, including speech recognition, noise reduction in communication channels, re-mixing of recorded music, and using environmental sound in musical compositions and film scores, as well as applications in environmental noise control. Many approaches to the problem have been investigated, each with application in a specific area. This paper presents a novel approach that would have application where: a) high-quality reproduction is desired with minimum artifacts; b) measurement using a multiple-microphone array is possible; and c) real-time performance is not required. As such it would apply particularly to audio-oriented applications, but may also have application in environmental noise. The technique involves a constrained least-squares decomposition of spectrogram values recorded at multiple microphones, together with an optional adaptive filtering step. Performance of the a...
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A real-time method of string instrument acoustic transfer which includes damping is proposed. Acoustic transfer of string instruments is relevant when trying to make a non-resonant instrument, such as an electric guitar, sound more... more
A real-time method of string instrument acoustic transfer which includes damping is proposed. Acoustic transfer of string instruments is relevant when trying to make a non-resonant instrument, such as an electric guitar, sound more similar to an acoustic guitar. Unlike previous acoustic transfer methods which only perform equalization, this method includes damping changes by using a time-varying filter which adds frequency-dependent exponential damping. Efficient digital filters are fit to bridge admittance measurements of an acoustic guitar and used to create equalization filters as well as damping correction filters. The damping correction filters work in real-time as they are triggered by onset and pitch detection of the signal measured through an under saddle pickup to determine the intensity of the damping.
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Due to geological closures between 21 000 and 29 000 years ago, the acoustics of the UNESCO World Heritage site, Chauvet Cave (Ardèche, France) have been in slow flux via mineral deposition processes that continue to alter its interior.... more
Due to geological closures between 21 000 and 29 000 years ago, the acoustics of the UNESCO World Heritage site, Chauvet Cave (Ardèche, France) have been in slow flux via mineral deposition processes that continue to alter its interior. Since Upper Paleolithic humans created extensive and elaborate artworks throughout this grand limestone cavern more than 30 000 years ago, the cave’s interior has changed with calcite and other minerals forming a diversity of features including the best-known stalactites, stalagmites, and flowstone floor coverings. Here, we report on archaeoacoustics fieldwork in 2022 that initiated acoustical mapping and reconstructive modeling to enable archaeological acoustics research and the creation of auralizations and multimodal experiences for virtual public access to this conservation-restricted place. We present here a comparative room acoustics study of two substantively enclosed cave areas (Salle du Fond and Galerie du Cactus) whose volumes differ significantly, but whose extant reverberation times are similar across most center bands, providing important information about the dynamical contributions of surface materials and structural features distinct to each impulse-response-measured location. Our study exemplifies an archaeological application of room acoustics methods with site-responsive techniques that offer a human-centered approach for understanding and translating cultural heritage acoustics across time.
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When a monophonic source signal is projected from two or more loudspeakers, listeners typically perceive a single, phantom source, positioned according to the relative signal amplitudes and speaker locations. While this property is the... more
When a monophonic source signal is projected from two or more loudspeakers, listeners typically perceive a single, phantom source, positioned according to the relative signal amplitudes and speaker locations. While this property is the basis of modern panning algorithms, it is often desirable to control the perceived spatial extent of the phantom source, or to project multiple, separately perceived copies of the signal. So that the human auditory system does not process the loudspeaker outputs as a single coherent source, these effects are commonly achieved by generating a set of mutually decorrelated (e.g., statistically independent) versions of the source signal, which are then panned to make an extended source or multiple, independent source copies. In this paper, we introduce an approach to decorrelation using randomly generated allpass filters, and introduce numerical methods for evaluating the perceptual effectiveness of decorrelation algorithms. By using allpass filters, the ...
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We present an analysis of the cowbell voice circuit from the Roland TR-808 Rhythm Composer. A digital model based on this analysis accurately emulates the original. Through the use of physical and behavioral models of each sub-circuit,... more
We present an analysis of the cowbell voice circuit from the Roland TR-808 Rhythm Composer. A digital model based on this analysis accurately emulates the original. Through the use of physical and behavioral models of each sub-circuit, this model supports accurate emulation of circuit-bent extensions to the voice's original behavior (including architecture-level alterations and component substitution). Some of this behavior is very complicated and is inconvenient or impossible to capture accurately through black box modeling or structured sampling. The band pass filter sub-circuit is treated as a case study of how to apply Mason's gain formula to finding the continuous-time transfer function of an analog circuit.
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ABSTRACT One dimensional digital waveguides are widely used to model travelling pressure waves along wind instrument bores. These models must also account for frequency-dependent losses occurring along the bore walls, and at boundaries.... more
ABSTRACT One dimensional digital waveguides are widely used to model travelling pressure waves along wind instrument bores. These models must also account for frequency-dependent losses occurring along the bore walls, and at boundaries. This is accomplished by incorporating waveguide filter elements, which are based on the widely accepted theory describing these losses. A measurement technique is demonstrated that allows the effects of each waveguide element to be isolated and observed for simple cylindrical and conical tube structures, as well as their combination. This measurement system yields data that closely matches the theory and provides confidence that it may be extended to accurately measure musical instrument bores, where bore shapes are usually considerably less simple, and thus more difficult to account for theoretically. This work paves the way for further application to clarinet and trumpet bores.
A reed, or more generally, a pressure-controlled valve, is the primary resonator for many wind instruments and vocal systems. In physical modeling synthesis, the method used for simulating the reed typically depends on whether an... more
A reed, or more generally, a pressure-controlled valve, is the primary resonator for many wind instruments and vocal systems. In physical modeling synthesis, the method used for simulating the reed typically depends on whether an additional upstream or downstream pressure causes the corresponding side of the valve to open or close further. In this work, a generalized and configurable model of a pressure controlled valve is presented, allowing the user t o design a reed simply by setting the model parameters. The parameters are continuously variable, and may be config- ured to produce blown closed models (like woodwinds or reed-pipes), blown open models (as in simple lip-reeds, the human larynx, harmonicas and harmoniums) and sym- metric "swinging door" models. This generalized virtual reed affords the musician the ability to produce a wide va- riety of sounds which would otherwise only be obtained with several reed instruments.
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The control of virtual musical instruments often relies on either a specially-developed controller on which the performer has usually not gained sufficient virtuosity to play musically, or an existing multipurpose general controller with... more
The control of virtual musical instruments often relies on either a specially-developed controller on which the performer has usually not gained sufficient virtuosity to play musically, or an existing multipurpose general controller with control parameters not always easily, or intuitively, mapped to the synthesis parameters of the virtual instrument being performed. A response to this problem is to obtain control information from a musical performance where the performer uses an instrument with which s/he is sufficiently familiar. In this work, we incorporate a previously developed measurement technique to transform a measured clarinet signal into a sequence of pulses corresponding to the reed displacement as a function of time. The measurement technique, shown to obtain accurate reflection functions from various tube structures, is used to obtain a filter modeling the bore and bell of the wind instrument used in the performance. The "reed pulse" waveform is then isolated...
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In this work we present a technique for estimating the reed flow signal, typically a periodic sequence of pulses, from the recorded sound of a reed instrument. The instrument is modeled as a reed coupled to a 1-D waveguide having unknown... more
In this work we present a technique for estimating the reed flow signal, typically a periodic sequence of pulses, from the recorded sound of a reed instrument. The instrument is modeled as a reed coupled to a 1-D waveguide having unknown filter elements that must first be determined before constructing the instrument reed flow transfer function. As pressure waves make two round trips from the mouthpiece to the bell and back for each reed pulse, the output periodic pressure has two distinct halves: the second half being roughly the first half filtered by the instrument’s propagation losses. Estimation of these losses is not simply a spectral ratio, as the two halves are not temporally disjoint and the beginning of the reed pulse period is often unclear. The running autocorrelation of the recorded signal is zero phase and naturally provides the beginning of the period of the recorded signal as well as clear first and second phases that may be analyzed to estimate the round-trip losses...
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In this work, a technique is presented for estimating the reed pulse from the pressure signal recorded at the bell of a clarinet during performance. The reed pulse is a term given to the typically periodic sequence of bore input pressure... more
In this work, a technique is presented for estimating the reed pulse from the pressure signal recorded at the bell of a clarinet during performance. The reed pulse is a term given to the typically periodic sequence of bore input pressure pulses, a signal related to the volume flow through a vibrating reed by the characteristic impedance of the aperture to the bore. The problem is similar to extracting glottal pulse sequence from recorded speech; however, because the glottis and instrument reeds have very different masses and opening areas, the source-filter model used in speech processing is not applicable. Here, the reed instrument is modeled as a pressure-controlled valve coupled to a bi-directional waveguide, with the output pressure approximated as a linear time invariant transformation of the product of reed volume flow and the characteristic impedance of the bore. By noting that pressure waves will make two round trips from the mouthpiece to the bell and back for each reed pul...
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In this work, a method is presented for estimating the reflection off the clarinet mouthpiece, using a priori measurement of the bell, and post processing of the instrument's produced sound. A previously introduced measurement... more
In this work, a method is presented for estimating the reflection off the clarinet mouthpiece, using a priori measurement of the bell, and post processing of the instrument's produced sound. A previously introduced measurement technique is used to obtain measurement of clarinet bell and transmission filters. In addition to these elements, however, the round-trip propagation loss in the clarinet bore and bell also includes wall loss and mouthpiece reflection. Though the former is accurately modeled theoretically, assuming the clarinet bell is close to cylindrical, the mouthpiece is more difficult to measure, both because of a supposed oscillating reed, and because the required placement of a measurement device would obstruct the mouthpiece's characteristic reflection. The lumped round-trip loss filter in the bore is estimated from the clarinet signal by first considering the signal's periodic structure. After taking the signal's autocorrelation, which preserves its pe...
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In computer music, a digital simulation of the pressure-controlled valve is required to model many wind instruments and vocal systems. The method used for simulating the valve usually depends on how closely it corresponds to any of three... more
In computer music, a digital simulation of the pressure-controlled valve is required to model many wind instruments and vocal systems. The method used for simulating the valve usually depends on how closely it corresponds to any of three simple configurations for valves in acoustic tubes. The classification of the valve is determined by whether an additional upstream or downstream pressure causes the valve to open or close further on the corresponding side. In this work, a configurable model of a pressure controlled valve is implemented, allowing the user to design a valve simply by changing the model parameters. The user may design three simple models: one that is blown closed (like woodwinds or reed-pipes), one that is blown open (as in simple lip-reeds, the human larynx, harmonicas and harmoniums) and the transverse model (as in the bird’s vocal organ, the syrinx). Using the computer model, the musical applications of different valve configurations and topologies (where valves ar...
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Pressure controlled valves are the primary sound production mechanisms for woodwind and brass musical instruments, as well as for many bioacoustic vocal systems such as the larynx and syrinx (the vocal organ in birds). During sound... more
Pressure controlled valves are the primary sound production mechanisms for woodwind and brass musical instruments, as well as for many bioacoustic vocal systems such as the larynx and syrinx (the vocal organ in birds). During sound production, air flow sets a reed or membrane into motion creating a variable height in the valve channel and, potentially, periodically closing the channel completely. Depending on how this event is handled, an abrupt termination of air flow between open and closed states can cause undesirable discontinuities and inaccuracies in a discrete-time simulation—particularly at relatively low audio sampling rates. A solution was developed by re-examining the behavior of the differential equation governing volume flow through a pressure-controlled valve, paying particular attention to this rather troublesome transition. A closed-form solution for the time evolution of volume flow is given and used to derive an update for volume flow. The result is a smoother, mor...
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In this work we estimate the volume flow pulses through a clarinet reed from recorded clarinet signal. The idea is similar to extracting glottal pulse sequences from recorded speech, however since the clarinet reed has little mass and... more
In this work we estimate the volume flow pulses through a clarinet reed from recorded clarinet signal. The idea is similar to extracting glottal pulse sequences from recorded speech, however since the clarinet reed has little mass and generates significant ...
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Presented of the 6th International Conference on Auditory Display (ICAD), Atlanta, GA, April 2-5, 2000This paper discusses development issues for a software-based, real-time virtual audio rendering system, Sound Lab (SLAB), designed to... more
Presented of the 6th International Conference on Auditory Display (ICAD), Atlanta, GA, April 2-5, 2000This paper discusses development issues for a software-based, real-time virtual audio rendering system, Sound Lab (SLAB), designed to work in the personal computer environment using a standard signal-processing library. The system, which is being developed as a tool for the study of spatial hearing, takes advantage of the low-cost PC platform while providing a flexible, maintainable, and extensible architecture to enable the quick development of experiments. The current capabilities and dynamic behavior of the SLAB system are described
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This paper presents the Tibetan Singing Prayer Wheel, a hand-held, wireless, sensor-based musical instrument with a human-computer interface that simultaneously processes vocals and synthesizes sound based on the performer's hand... more
This paper presents the Tibetan Singing Prayer Wheel, a hand-held, wireless, sensor-based musical instrument with a human-computer interface that simultaneously processes vocals and synthesizes sound based on the performer's hand gestures with a one-to-many mapping strategy. A physical model simulates the singing bowl, while a modal reverberator and a delay-and-window effect process the performer's vocals. This system is designed for an electroacoustic vocalist interested in using a solo instrument to achieve performance goals that would normally require multiple instruments and activities.
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This Convention paper was selected based on a submitted abstract and 750-word precis that have been peer reviewed by at least two qualified anonymous reviewers. The complete manuscript was not peer reviewed. This convention paper has been... more
This Convention paper was selected based on a submitted abstract and 750-word precis that have been peer reviewed by at least two qualified anonymous reviewers. The complete manuscript was not peer reviewed. This convention paper has been reproduced from the author's advance manuscript without editing, corrections, or consideration by the Review Board. The AES takes no responsibility for the contents. Additional papers may be obtained by sending request and remittance to Audio Engineering Society, 60 East 42 nd Street ABSTRACT We present a computational acoustic model of the well-preserved interior architecture at the 3,000-year-old Andean ceremonial center at Chavín de Huántar, Perú. Our previous model prototype [Kolar et al. 2010] translated the acoustically coupled topology of Chavín gallery forms to a model based on digital waveguides (bi-directional by definition), representing passageways, connected through reverberant scattering junctions, representing the larger room-lik...
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Measurements of acoustic input impedance of wind instruments using two different approaches are presented. In the first approach, a tube is connected to the instrument and excited with broad-band noise. Signals recorded at microphone... more
Measurements of acoustic input impedance of wind instruments using two different approaches are presented. In the first approach, a tube is connected to the instrument and excited with broad-band noise. Signals recorded at microphone pairs placed along the tube are then analyzed to estimate the instrument input impedance. A calibration step is described, wherein the position of each microphone pair is determined from the measurement of a rigid termination. The second technique makes use of a long tube with a single microphone located at its midpoint. Using a swept sinusoid stimulus, the impulse response is measured for the tube, first with a rigid termination, and then with the system to be characterized attached. The system reflectance, and therefore its impedance, is found by comparing the first reflection from the tube end for both measurements. The design of the impedance probes and the data sampling and analysis procedures are presented. Measurements obtained using the two tech...
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The diode clipper circuit with an embedded low-pass filter lies at the heart of both diode clipping “Distortion ” and “Overdrive ” or “Tube Screamer ” effects pedals. An accurate simulation of this circuit requires the solution of a... more
The diode clipper circuit with an embedded low-pass filter lies at the heart of both diode clipping “Distortion ” and “Overdrive ” or “Tube Screamer ” effects pedals. An accurate simulation of this circuit requires the solution of a nonlinear ordinary differential equation (ODE). Numerical methods with stiff stability – Backward Euler, Trapezoidal Rule, and second-order Backward Difference Formula – allow the use of relatively low sampling rates at the cost of accuracy and aliasing. However, these methods require iteration at each time step to solve a nonlinear equation, and the tradeoff for this complexity must be evaluated against simple explicit methods such as Forward Euler and fourth order Runge-Kutta, which require very high sampling rates for stability. This paper surveys and compares the basic ODE solvers as they apply to simulating circuits for audio processing. These methods are compared to a static nonlinearity with a pre-filter. It is found that implicit or semiimplicit ...
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Measurements of acoustic input impedance of wind instruments using two different approaches are presented. In the first approach, commonly referred to as the two-microphone transfer function method, a tube is connected to the instrument... more
Measurements of acoustic input impedance of wind instruments using two different approaches are presented. In the first approach, commonly referred to as the two-microphone transfer function method, a tube is connected to the instrument and excited with broad-band noise. Signals recorded at microphone pairs placed along the tube are then analyzed to estimate the instrument input impedance. A calibration step is described, wherein the position of each microphone pair is determined from the measurement of a rigid termination. The second technique, a novel variant of pulse reflectometry, makes use of a long tube with a single microphone located at its midpoint. Using a long-duration broad-band stimulus, the impulse response is measured for the tube, first with a rigid termination, and then with the system to be characterized attached. The system reflectance, and therefore its impedance, is found by comparing the first reflection from the tube end for both measurements. The design of th...
The problem of a self-calibrating speaker and microphone array is considered. Each speaker in the array is assumed to have local processing power and the ability to communicate acous-tically with all other speakers and microphones. This... more
The problem of a self-calibrating speaker and microphone array is considered. Each speaker in the array is assumed to have local processing power and the ability to communicate acous-tically with all other speakers and microphones. This paper describes a preliminary high-level system design, including a proposed algorithm for computing array geometry from measured inter-speaker and speaker-to-microphone distances. 1
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Pitch glide is an important effect that occurs in nearly all plucked string instruments. In essence, large amplitude waves traveling on a string during the note onset increases the string tension above its nominal value, and therefore... more
Pitch glide is an important effect that occurs in nearly all plucked string instruments. In essence, large amplitude waves traveling on a string during the note onset increases the string tension above its nominal value, and therefore cause the pitch to temporarily in-crease. Measurements are presented showing an exponential re-laxation of all the partial frequencies to their nominal values with a time-constant related to the decay rate of transverse waves prop-agating on the string. This exponential pitch trajectory is sup-ported by a simple physical model in which the increased tension is somewhat counterbalanced by the increased length of the string. Finally, a method for synthesizing the plucked string via a novel hybrid digital waveguide-modal synthesis model is presented with implementation details for time-varying resonators. 1.
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Cappella Romana (CR420-CDBR, 2019). Booklet essays by Bissera V. Pentcheva, Jonathan S. Abel and Elliot K. Canfield Dafilou, and Alexander Lingas. Includes CD and Blu-ray discs. The Blu-ray includes the documentary film 'The Voice... more
Cappella Romana (CR420-CDBR, 2019). Booklet essays by Bissera V. Pentcheva, Jonathan S. Abel and Elliot K. Canfield Dafilou, and Alexander Lingas. Includes CD and Blu-ray discs. The Blu-ray includes the documentary film 'The Voice of Hagia Sophia', directed, edited and co-produced by Duygu Eruçman; Produced by Bissera V. Pentcheva. Cinematography by Meryem Yavuz, Michael Seely, and Ben Wu. Part of the Icons of Sound project: https://ccrma.stanford.edu/groups/iconsofsound/
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An algorithm for artistic spectral audio processing and synthesis using allpass filters is presented. These filters express group delay trajectories, allowing fine control of their frequency-dependent arrival times. We present methods for... more
An algorithm for artistic spectral audio processing and synthesis using allpass filters is presented. These filters express group delay trajectories, allowing fine control of their frequency-dependent arrival times. We present methods for designing the group delay trajectories to yield a novel class of filters for sound synthesis and audio effects processing. A number of categories of group delay trajectory design are discussed, including stair-stepped, modulated, and probabilistic. Synthesis and processing examples are provided.
A simple, robust method for measuring echo density from a reverberation impulse response is presented. Based on the property that a reverberant field takes on a Gaussian distribution once an acoustic space is fully mixed, the measure... more
A simple, robust method for measuring echo density from a reverberation impulse response is presented. Based on the property that a reverberant field takes on a Gaussian distribution once an acoustic space is fully mixed, the measure counts samples lying outside a standard deviation in a given impulse response window and normalizes by that expected for Gaussian noise. The measure is insensitive to equalization and reverberation time, and is seen to perform well on both artificial reverberation and measurements of room impulse responses. Listening tests indicate a correlation between echo density measured in this way and perceived temporal quality or texture of the reverberation.
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Notebook interfaces in computing, introduced in the late 1980s, are in active modern use by data science and machine learning communities. Related to literate computing, notebooks encourage interleaving expository text with data, code,... more
Notebook interfaces in computing, introduced in the late 1980s, are in active modern use by data science and machine learning communities. Related to literate computing, notebooks encourage interleaving expository text with data, code, and figures, making for intuitive presentation of results. During development, they allow for nonlinear or exploratory development, and encourage building on prior research. We consider the application of such notebooks in audio plugin development and analysis, providing short example notebooks covering scenarios in DSP tutorials, white-box testing, blackbox testing, and automation of third-party tools. While noting these workflows have been supported by commercial tools for decades, we exclusively use a range of FOSS languages and tools in our samples.
The mixing matrix of a Feedback Delay Network (FDN) reverberator is used to control the mixing time and echo density profile. In this work, we investigate the effect of the mixing matrix on the modes (poles) of the FDN with the goal of... more
The mixing matrix of a Feedback Delay Network (FDN) reverberator is used to control the mixing time and echo density profile. In this work, we investigate the effect of the mixing matrix on the modes (poles) of the FDN with the goal of using this information to better design the various FDN parameters. We find the modal decomposition of delay network reverberators using a state space formulation, showing how modes of the system can be extracted by eigenvalue decomposition of the state transition matrix. These modes, and subsequently the FDN parameters, can be designed to mimic the modes in an actual room. We introduce a parameterized orthonormal mixing matrix which can be continuously varied from identity to Hadamard. We also study how continuously varying diffusion in the mixing matrix affects the damping and frequency of these modes. We observe that modes approach each other in damping and then deflect in frequency as the mixing matrix changes from identity to Hadamard. We also qu...
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This research presents a uniform approximation to the formulas of Benade and Keefe for the propagation constant of a cylindrical tube, valid for all tube radii and frequencies in the audio range. Based on this approximation, a simple... more
This research presents a uniform approximation to the formulas of Benade and Keefe for the propagation constant of a cylindrical tube, valid for all tube radii and frequencies in the audio range. Based on this approximation, a simple expression is presented for a filter which closely matches the thermoviscous loss filter of a tube of specified length and radius at a given sampling rate. The form of this filter and the simplicity of coefficient calculation make it particularly suitable for real-time music applications where it may be desirable to have tube parameters such as length and radius vary during performance.
Research Interests: Mathematics and Real Time
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... more
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 effi...
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The short-time Fourier transform (STFT) based spectrogram is commonly used to analyze the time-frequency content of a signal. Depending on window size, the STFT provides a trade-off between time and frequency resolutions. This paper... more
The short-time Fourier transform (STFT) based spectrogram is commonly used to analyze the time-frequency content of a signal. Depending on window size, the STFT provides a trade-off between time and frequency resolutions. This paper presents a novel method that achieves high resolution simultaneously in both time and frequency. We extend Probabilistic Latent Component Analysis (PLCA) to jointly decompose two spectrograms, one with a high time resolution and one with a high frequency resolution. Using this decomposition, a new spectrogram, maintaining high resolution in both time and frequency, is constructed. Termed the “super-resolution spectrogram”, it can be particularly useful for speech as it can simultaneously resolve both glottal pulses and individual harmonics.
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Dispersive delay and comb filters, implemented as a parallel sum of high-Q mode filters tuned to provide a desired frequency-dependent delay characteristic, have advantages over dispersive filters that are implemented using cascade or... more
Dispersive delay and comb filters, implemented as a parallel sum of high-Q mode filters tuned to provide a desired frequency-dependent delay characteristic, have advantages over dispersive filters that are implemented using cascade or frequency-domain architectures. Here we present techniques for designing the modal filter parameters for music and audio applications. Through examples, we show that this parallel structure is conducive to interactive and time-varying modifications, and we introduce extensions to the basic model.
In previous work, the authors presented a generalized parametric model of a pressure controlled valve, allowing the user to design a continuum of reed configurations, including “blown open”, “blown closed” and the “swinging door”. Though... more
In previous work, the authors presented a generalized parametric model of a pressure controlled valve, allowing the user to design a continuum of reed configurations, including “blown open”, “blown closed” and the “swinging door”. Though the generalized reed model behaved as expected, the quality of the produced sound was somewhat limited, likely due to the dependence of reed oscillation on the connected instrument bore and bell. In this work we further explore the sound production of the generalized reed by incorporating reflection filters measured from actual musical instruments. The measurement technique is shown to produce results closely matching theoretical expectation for cylindrical and conical tubes, and is applied to the clarinet and trumpet. Measurements are incorporated into a waveguide model using the generalized reed.
Research Interests:
This paper describes methods for processing signals recorded at a microphone array so as to estimate the signals that would have appeared at the elements of a different, collocated microphone array, i.e., “translating” measurements made... more
This paper describes methods for processing signals recorded at a microphone array so as to estimate the signals that would have appeared at the elements of a different, collocated microphone array, i.e., “translating” measurements made at one microphone array to those hypothetically appearing at another array. Two approaches are proposed; a non-parametric method in which a fixed, low-sidelobe beamformer applied to the “source” array drives virtual sources rendered on the “target” array, and a parametric technique in which constrained beamformers are used to estimate source directions, with the sources extracted and rendered to the estimated directions. Finally, a hybrid method is proposed, which combines both approaches so that the extracted point sources and residual can be separately rendered. Experimental results using an array of 2mm-diameter microphones and human HRTFs are reported as a simple example.
Hilbert Transformers have found many signal processing applications, from single-sideband communication systems to audio effects. IIR implementations are attractive for computational efficiency. In this paper, we present a complete design... more
Hilbert Transformers have found many signal processing applications, from single-sideband communication systems to audio effects. IIR implementations are attractive for computational efficiency. In this paper, we present a complete design procedure for an efficient infinite impulse response (IIR) Hilbert transformer filter. We start from a half-band filter design, and show how the poles move as the half-band filter is transformed into summed all-pass filters and then into a Hilbert transformer filter. The design technique is based entirely on pole locations, and creates a numerically robust filter in cascaded first-order allpass form.
This paper describes a sound synthesis technique th at modulates the coefficients of allpass filter chains using aud io-rate frequencies. It was found that modulating a single allpass filter section produces a feedback AM‐like spectrum,... more
This paper describes a sound synthesis technique th at modulates the coefficients of allpass filter chains using aud io-rate frequencies. It was found that modulating a single allpass filter section produces a feedback AM‐like spectrum, and that its bandwidth is extended and further processed by non-sinusoidal FM when the sections are cascaded. The cascade length parameter provides dynamic bandwidth control to prevent upper range al iasing artifacts, and the amount of spectral content within th at band can be controlled using a modulation index parameter. The technique is capable of synthesizing rich and evolving timbres, including those resembling classic virtual analog waveforms. It can also be used as an audio effect with pitch-tracked input so urces. Software and sound examples are available at http://www.acoustics.hut.fi/publications/papers/dafx09 -cm/
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We designed models of three TR-808 voice circuits which are physically-informed and derived as closed-form expressions in terms of electrical component values where possible. An important part of the performance practice of analog drum... more
We designed models of three TR-808 voice circuits which are physically-informed and derived as closed-form expressions in terms of electrical component values where possible. An important part of the performance practice of analog drum machines involves modifying their voices circuits. These techniques would be impossible to recreate in a digital model through structured sampling or black-box modeling. Our physically-informed models are designed to respect this practice by allowing changes to the circuit topology and electrical component values. Ad-hoc simplifications reduce the models' computational load without affecting perceptual accuracy.
Research Interests:
Invited paper presented at 2nd Pan American/Iberian Meeting on Acoustics, Cancún, México, November 2010. ABSTRACT In 2001, twenty Strombus galeatus marine shell trumpets were excavated at the 3,000 year-old ceremonial center at... more
Invited paper presented at 2nd Pan American/Iberian Meeting on Acoustics, Cancún, México, November 2010.
ABSTRACT
In 2001, twenty Strombus galeatus marine shell trumpets were excavated at the 3,000 year-old ceremonial center at Chavín de Huántar, Perú, marking the first documented contextual discovery of intact sound-producing instruments at this Formative Period site in the Andean highlands. These playable shells are decorated and crafted for musical use with well-formed mouthpieces created by cutting the small end (spire) off and grinding/polishing the resulting opening. The shells are use-polished, and additionally modified with a v-shaped cut to the outer apical lip. We present an acoustic analysis of the measured response of each instrument, to a variety of excitations, at microphones placed in the mouthpiece, player's mouth, bore, bell, and surrounding near-field. From these measurements we characterize each instrument's sounding frequencies (fundamental and 1st overtone where possible), radiation pattern, and impedance, and we estimate the bore area function of each shell. Knowledge of the specific acoustic capabilities of these pututus allows us to understand and test their potential as sound sources in the ancient Chavín context, whose architectural acoustics are simultaneously studied by our research group.
ABSTRACT
In 2001, twenty Strombus galeatus marine shell trumpets were excavated at the 3,000 year-old ceremonial center at Chavín de Huántar, Perú, marking the first documented contextual discovery of intact sound-producing instruments at this Formative Period site in the Andean highlands. These playable shells are decorated and crafted for musical use with well-formed mouthpieces created by cutting the small end (spire) off and grinding/polishing the resulting opening. The shells are use-polished, and additionally modified with a v-shaped cut to the outer apical lip. We present an acoustic analysis of the measured response of each instrument, to a variety of excitations, at microphones placed in the mouthpiece, player's mouth, bore, bell, and surrounding near-field. From these measurements we characterize each instrument's sounding frequencies (fundamental and 1st overtone where possible), radiation pattern, and impedance, and we estimate the bore area function of each shell. Knowledge of the specific acoustic capabilities of these pututus allows us to understand and test their potential as sound sources in the ancient Chavín context, whose architectural acoustics are simultaneously studied by our research group.
Research Interests:
Invited paper presented at Acoustics '08, Paris, France, July 2008. ABSTRACT Chavín de Huántar is a monumental World Heritage archaeological site in the Peruvian highlands, predating Inca society by over 2000 years. The importance... more
Invited paper presented at Acoustics '08, Paris, France, July 2008.
ABSTRACT
Chavín de Huántar is a monumental World Heritage archaeological site in the Peruvian highlands, predating Inca society by over 2000 years. The importance of site acoustics is suggested by distinctive architectural features, notably an extensive network of underground galleries used in part for ritual purposes. The labyrinthine galleries are stone-walled and arranged in a series of small rectangular alcoves off narrow corridors. In this work, we initiate research that seeks to understand how the acoustics at Chavín may have influenced auditory experience.
Acoustic measurements and models of a site can be used to archive site acoustics, estimate the acoustics of inaccessible or alternative site architectures, and reconstruct the acoustics of modified or damaged sectors; they may also corroborate aspects of rituals suggested by other archaeological data. Preliminary measurements at Chavín show a short reverberation time, dense and energetic early reflections, and low inter-aural cross correlation. The short reverberation time would enable rhythmically articulated playing of Strombus shell trumpets found on site. The early reflection patterns would provide strong acoustic reinforcement and resonances in gallery alcoves. The wide soundfields would provide a sense of spaciousness and envelopment, contributing to ritual experience.
ABSTRACT
Chavín de Huántar is a monumental World Heritage archaeological site in the Peruvian highlands, predating Inca society by over 2000 years. The importance of site acoustics is suggested by distinctive architectural features, notably an extensive network of underground galleries used in part for ritual purposes. The labyrinthine galleries are stone-walled and arranged in a series of small rectangular alcoves off narrow corridors. In this work, we initiate research that seeks to understand how the acoustics at Chavín may have influenced auditory experience.
Acoustic measurements and models of a site can be used to archive site acoustics, estimate the acoustics of inaccessible or alternative site architectures, and reconstruct the acoustics of modified or damaged sectors; they may also corroborate aspects of rituals suggested by other archaeological data. Preliminary measurements at Chavín show a short reverberation time, dense and energetic early reflections, and low inter-aural cross correlation. The short reverberation time would enable rhythmically articulated playing of Strombus shell trumpets found on site. The early reflection patterns would provide strong acoustic reinforcement and resonances in gallery alcoves. The wide soundfields would provide a sense of spaciousness and envelopment, contributing to ritual experience.
Research Interests:
Research presented at the Audio Engineering Society 133rd Convention, San Francisco, CA, 26-29 October 2012. ABSTRACT We present a computational acoustic model of the well-preserved interior architecture at the 3,000-year-old Andean... more
Research presented at the Audio Engineering Society 133rd Convention, San Francisco, CA, 26-29 October 2012.
ABSTRACT
We present a computational acoustic model of the well-preserved interior architecture at the 3,000-year-old Andean ceremonial center Chavín de Huántar, Perú. Our previous model prototype [Kolar et al. 2010] translated the acoustically coupled topology of Chavín gallery forms to a model based on digital waveguides (bi-directional by definition), representing passageways, connected through reverberant scattering junctions, representing the larger room-like areas. Our new approach treats all architectural units as "reverberant" digital waveguides, with scattering junctions at the discrete planes defining the unit boundaries. In this extensible and efficient lumped-element model, we combine architectural dimensional and material data with sparsely measured impulse responses to simulate multiple and circulating arrival paths between sound sources and listeners.
ABSTRACT
We present a computational acoustic model of the well-preserved interior architecture at the 3,000-year-old Andean ceremonial center Chavín de Huántar, Perú. Our previous model prototype [Kolar et al. 2010] translated the acoustically coupled topology of Chavín gallery forms to a model based on digital waveguides (bi-directional by definition), representing passageways, connected through reverberant scattering junctions, representing the larger room-like areas. Our new approach treats all architectural units as "reverberant" digital waveguides, with scattering junctions at the discrete planes defining the unit boundaries. In this extensible and efficient lumped-element model, we combine architectural dimensional and material data with sparsely measured impulse responses to simulate multiple and circulating arrival paths between sound sources and listeners.