Digital Audio Effects

AudioSculpt is an application for the musical analysis and processing of sound files. The program unites a very detailed inspection of sound, both visually and auditorily, with high quality analysis-driven effects, such as timestretch, transposition and spectral filtering. Multiple algorithms provide automatic segmentation to guide the placement of sound treatments and steer processing parameters. By designing transformations directly on the sonogram, very precise spectral modifications can be made, allowing both intuitive sound design as well as sound restoration and source separation.

A physical model has been developed for real-time sound synthesis of the Clavinet, an electromechanical keyboard instrument from the 20th century. The Clavinet has a peculiar excitation mechanism, relying on a tangent striking the string. The modeling paradigm chosen is waveguide synthesis and this paper suggests several novel techniques, such as a polynomial excitation pulse model and a beating generator, both of which have parameters depending on key velocity. Realistic emulation of the release part of Clavinet tones is based on a decrease in the decay rate of the tone and lengthening of a delay line, which corresponds to the physical string. A real-time implementation on Pure Data demonstrates the efficiency of proposed model.

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 solvers are preferred and that the filter/static nonlinearity approximation is often perceptually adequate.

Procedural audio may be defined as real-time sound generation according to programmatic rules and live input. It is often considered a subset of sound synthesis and is especially applicable to nonlinear media, such as video games, virtual reality experiences and interactive audiovisual installations. However, there is resistance to widespread adoption of procedural audio because there is little awareness of the state of the art, including the diversity of sounds that may be generated, the controllability of procedural audio models, and the quality of the sounds that it produces. The authors address all of these aspects in this review paper, while attempting a largescale categorization of sounds that have been approached through procedural audio techniques. The role of recent advancements in neural audio synthesis, its current implementations, and potential future applications in the field are also discussed. Review materials are available * .

An impulse response of an enclosed reverberant space is composed of three basic components: the direct sound, early reflections and late reverberation. While the direct sound is a single event that can be easily identified, the division between the early reflections and late reverberation is less obvious as there is a gradual transition between the two. This paper explores two statistical measures that can aid in determining a point in time where the early reflections have transitioned into late reverberation. These metrics exploit the similarities between late reverberation and Gaussian noise that are not commonly found in early reflections. Unlike other measures, these need no prior knowledge about the rooms such as geometry or volume.

This paper is concerned with the design of efficient and controllable filters for sound synthesis purposes, in the context of the generation of sounds radiated by nonlinear sources. These filters are coupled and generate tonal components in an interdependent way, and are intended to emulate realistic perceptually salient effects in musical instruments in an efficient manner. Control of energy transfer between the filters is realized by defining a matrix containing the coupling terms. The generation of prototypical sounds corresponding to nonlinear sources with the filter bank is presented. In particular, examples are proposed to generate sounds corresponding to impacts on thin structures and to the perturbation of the vibration of objects when it collides with an other object. The different sound examples presented in the paper and available for listening on the accompanying site tend to show that a simple control of the input parameters allows to generate sounds whose evocation is coherent, and that the addition of random processes allows to significantly improve the realism of the generated sounds.

Animals generate sounds for many biological functions from mating to finding food. The ability to use inexpensive, non-intrusive sensors to gain valuable insights about the biology of animals is a promising application area for information and communication (ICT) research. In this thesis, we focus on the acoustic classification of flying insects in support of entomological research related to public health monitoring and control. This thesis describes how we can develop smart sensors capable of monitoring insects to control their development as well as to analyze their habitat and identify which areas are most affected. The focus of this work was on mosquito detection, although we are interested in the general problem of flying insect detection, and identification. Mosquitos are the most common vector for disease-causing viruses and parasites. They are estimated to transmit various types of diseases to more than 700 million people annually in all continents. A prototype acoustic insect flight detector was built from an analog microphone coupled to a digital sound recorder capable of recording and providing real-time data as well as some environmental parameters over common wireless networks. The system was placed in several points of interest identified by the public health administration institute of Madeira Islands and already monitored using traditional mechanical traps. The sensors collected several hours of ambient sound recordings that were further analyzed to detect the feasibility of the acoustic sensing approach to detect and identify mosquitos. Our results show the practical feasibility of this low-cost, non-intrusive approach for monitoring mosquitos in places requiring vector monitoring aimed at mosquito control or eradication.

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.

An electromagnetic pickup capturing vibrations of steel strings in electric stringed instruments modifies the timbre of the instrument in various ways. The acoustic characteristics and modeling of the pickup position, the sensitivity width of the pickup, the dispersion of waves in a steel string, mixing options of several pickups, linear resonant filtering of the pickup circuit itself, and the distortion caused by the distance-dependent magnetic flux are studied.

In order to synthesize steel-stringed instruments, such as a guitar, a model of the pickup phenomenon is required. This model includes the pickup position, the sensitivity width of the transducer, mixing options with multiple pickups, linear resonant filtering, and distortion produced by the distance-dependent magnetic flux. A waveguide framework was used to describe frequency coloration of the pickup location and the low-pass effect of sensitivity width. The resulting models can be used in musical sound synthesis and digital effects. ...

This paper presents a high-quality real-time pitch-shifting algorithm with a time-varying factor for monophonic audio and musical signals. The pitch-shifting algorithm is based on the resampling and time-scale modification method. A new time-scale modification method has been developed which is called the Normalized Filtered Correlation Time-Scale Modification (NFC-TSM) method. It uses a ring buffer for time-scaling. The best splicing point is searched in the normalized low-pass filtered signal using the Average Magnitude ...

Unlike well accepted FPGA emulation for digital circuits, there is no winning emulation solution for analog and mixed-signal (AMS) circuits. This paper presents an analog circuit emulation based on wave digital filters (WDFs), which covers the entire flow of transforming an AMS circuit from SPICE netlist to hardware implementation in FPGA. More specifically, it presents the theoretical support of how to map linear and nonlinear circuit components to WDF. The detail implementation of each WDF component in FPGA is not elaborated due to the page limit. Experiments show that there is a virtually perfect match between FPGA emulation and HSPICE simulations on two small but representative analog circuits, indicating high accuracy of the proposed emulation, and the FPGA-based WDF emulation can process analog signal sampled at as high as 512KHz, which is adequate for a variety of biomedical sensing applications.

It is not uncommon to hear musicians and audio engineers speak of warmth and brightness when describing analog technologies such as vintage mixing consoles, multitrack tape machines, and valve compressors. What is perhaps less common, is hearing this term used in association with retro digital technology. A question exists as to how much the low bit rate and low-grade conversion quality contribute to the overall brightness or warmth of a sound when processed with audio effects simulating early sampling technologies. These two dimensions of timbre are notoriously difficult to define and more importantly, measure. We present a subjective user study of brightness and warmth, where a series of audio examples are processed with different audio effects. 26 participants rated the perceived level of brightness and warmth of various instrumental sequences for 5 different audio effects including bit depth reduction, compression and equalisation. Results show that 8 bit reduction tends to increase brightness and decrease warmth whereas 12 bit reduction tends to do the opposite, although this is very much dependent on the instrument. Interestingly, the most significant brightness changes, due to bit reduction, were obtained for bass sounds. For comparison purposes, instrument phrases were also processed with both an analogue compressor and an equalisation plugin to see if any subjective difference was noticed when simulating sonic characteristics that might be associated with warmth. Greater significance was observed when the sound excerpts were processed with the plugins being used to simulate the effects of bit depth reduction. CCS Concepts: • Information systems → Speech / audio search; • Applied computing → Sound and music computing; Media arts; • Computing methodologies → Perception; • Software and its engineering → Semantics.