Present systems for streaming digital audio between devices connected by internet have been limited by a number of compromises. Because of restricted bandwidth and “best effort” delivery, signal compression of one form or another is typical. Buffering of audio data which is needed to safeguard against delivery uncertainties can cause signal delays of seconds. Audio is in general an unforgiving test of networking, e.g., one data packet arriving too late and we hear it. Trade-offs of signal quality have been necessary to avoid this basic fact and until now, have vied against serious musical uses. Beginning in late 1998, audio applications specifically designed for next-generation networks were initiated that could meet the stringent requirements of professional-quality music streaming. A related experiment was begun to explore the use of audio as a network measurement tool. SoundWIRE (sound waves over the internet from real-time echoes) creates a sonar-like ping to display to the ear qualities of bidirectional connections. Recent experiments have achieved coast-to-coast sustained audio connections whose round trip times are within a factor of 2 of the speed of light. Full-duplex speech over these connections feels comfortable and in an IIR recirculating form that creates echoes like SoundWIRE, users can experience singing into a transcontinental echo chamber. Three simplifications to audio streaming are suggested in this paper: Compression has been eliminated to reduce delay and enhance signal-quality. TCP/IP is used in unidirectional flows for its delivery guarantees and thereby eliminating the need for application software to correct transmission errors. QoS puts bounds on latency and jitter affecting long-haul bidirectional flows.

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Digital waveguide (DW) modeling techniques are typically associated with a traveling-wave decomposition of wave variables and a “reflection function” approach to simulating acoustic systems. As well, it is often assumed that inputs and outputs to/from these systems must be formulated in terms of traveling-wave variables. In this paper, we provide a tutorial review of DW modeling of acoustic structures to show that they can easily accommodate physical input and output variables. Under certain constraints, these formulations reduce to simple “Schroeder reverb-like” computational structures. We also present a stable single-reed filter model that allows an explicit solution at the reed / air column junction. A clarinet-like system is created by combining the reed filter with a DW impedance model of a cylindrical air column.

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In this paper, the analysis and synthesis of a rolling ball sound is proposed. The approach is based on the assumption that the rolling sound is generated by a concatenation of micro-impacts between a ball and a surface, each having associated resonances. Contact timing information is first extracted from the rolling sound using an onset detection process. The resulting individual contact segments are subband filtered before being analyzed using linear predictive coding (LPC) and notch filter parameter estimation. The segments are then resynthesized and overlap-added to form a complete rolling sound. This approach is similar to that of [1], though the methods used for contact event detection and filter parameter estimation are completely different.

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In the context of efficient synthesis of wind instrument sound, we introduce a technique for joint modeling of input impedance and sound pressure radiation as digital filters in parallel form, with the filter coefficients derived from experimental data. In a series of laboratory measurements taken on an alto saxophone, the input impedance and sound pressure radiation responses were obtained for each fingering. In a first analysis step, we iteratively minimize the error between the frequency response of an input impedance measurement and that of a digital impedance model constructed from a parallel filter structure akin to the discretization of a modal expansion. With the modal coefficients in hand, we propose a digital model for sound pressure radiation which relies on the same parallel structure, thus suitable for coefficient estimation via frequency-domain least-squares. For modeling the transition between fingering positions, we propose a simple model based on linear interpolation of input impedance and sound pressure radiation models. For efficient sound synthesis, the common impedance-radiation model is used to construct a joint reflectanceradiation digital filter realized as a digital waveguide termination that is interfaced to a reed model based on nonlinear scattering.

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The hand harmonium is arguably the most popular instrument for vocal accompaniment in Hindustani music today. However, it lacks microtonality and the ability to produce controlled pitch glides, which are both important in Hindustani music. A harmonium sound synthesis model with a source-filter structure was previously presented by the authors in which the harmonium reed sound is synthesized using a physical model and the effect of the wooden enclosure is applied by a filter estimated from a recorded note. In this paper, we propose a simplified and perceptually informed signal model capable of real time synthesis with timbre control. In the signal model, the source is constructed as a band-limited waveform matching the spectral characteristics of the source signal in the physical model. Simplifications are suggested to parametrize the filter on the basis of prominent peaks in the filter frequency response. The signal model is implemented as a Pure Data [1] patch for live performance using a standard MIDI keyboard.

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