Download Group Delay-Based Allpass Filters for Abstract Sound Synthesis and Audio Effects Processing 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.
Download Joint modeling of impedance and radiation as a recursive parallel filter structure for efficient synthesis of wind instrument sound 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.
Download A Virtual Tube Delay Effect A virtual tube delay effect based on the real-time simulation of acoustic wave propagation in a garden hose is presented. The paper describes the acoustic measurements conducted and the analysis of the sound propagation in long narrow tubes. The obtained impulse responses are used to design delay lines and digital filters, which simulate the propagation delay, losses, and reflections from the end of the tube which may be open, closed, or acoustically attenuated. A study on the reflection caused by a finite-length tube is described. The resulting system consists of a digital waveguide model and produces delay effects having a realistic low-pass filtering. A stereo delay effect plugin in P URE DATA1 has been implemented and it is described here.
Download Musikverb: A Harmonically Adaptive Audio Reverberation We present MusikVerb, a novel digital reverberation capable of adapting its output to the harmonic context of a live music performance. The proposed reverberation is aware of the harmonic content of an audio input signal and ‘tunes’ the reverberation output to its harmonic content using a spectral filtering technique. The dynamic behavior of MusikVerb avoids the sonic clutter of traditional reverberation, and most importantly, fosters creative endeavor by providing new expressive and musically-aware uses of reverberation. Despite its applicability to any input audio signal, the proposed effect has been designed primarily as a guitar pedal effect and a standalone software application.
Download Real-Time Wave Digital Simulation of Cascaded Vacuum Tube Amplifiers using Modified Blockwise Method Vacuum tube amplifiers, known for their acclaimed distortion characteristics, are still widely used in hi-fi audio devices. However, bulky, fragile and power-consuming vacuum tube devices have also motivated much research on digital emulation of vacuum tube amplifier behaviors. Recent studies on Wave Digital Filters (WDF) have made possible the modeling of multi-stage vacuum tube amplifiers within single WDF SPQR trees. Our research combines the latest progress on WDF with the modified blockwise method to reduce the overall computational complexity of modeling cascaded vacuum tube amplifiers by decomposing the whole circuit into several small stages containing only two adjacent triodes. Certain performance optimization methods are discussed and applied in the eventual real-time implementation.
Download Stationary/transient Audio Separation Using Convolutional Autoencoders Extraction of stationary and transient components from audio has many potential applications to audio effects for audio content production. In this paper we explore stationary/transient separation using convolutional autoencoders. We propose two novel unsupervised algorithms for individual and and joint separation. We describe our implementation and show examples. Our results show promise for the use of convolutional autoencoders in the extraction of sparse components from audio spectrograms, particularly using monophonic sounds.
Download Improving intelligibility prediction under informational masking using an auditory saliency model The reduction of speech intelligibility in noise is usually dominated by energetic masking (EM) and informational masking (IM). Most state-of-the-art objective intelligibility measures (OIM) estimate intelligibility by quantifying EM. Few measures model the effect of IM in detail. In this study, an auditory saliency model, which intends to measure the probability of the sources obtaining auditory attention in a bottom-up process, was integrated into an OIM for improving the performance of intelligibility prediction under IM. While EM is accounted for by the original OIM, IM is assumed to arise from the listener’s attention switching between the target and competing sounds existing in the auditory scene. The performance of the proposed method was evaluated along with three reference OIMs by comparing the model predictions to the listener word recognition rates, for different noise maskers, some of which introduce IM. The results shows that the predictive accuracy of the proposed method is as good as the best reported in the literature. The proposed method, however, provides a physiologically-plausible possibility for both IM and EM modelling.
Download Modal Analysis Of Room Impulse Responses Using Subband Esprit This paper describes a modification of the ESPRIT algorithm which can be used to determine the parameters (frequency, decay time, initial magnitude and initial phase) of a modal reverberator that best match a provided room impulse response. By applying perceptual criteria we are able to match room impulse responses using a variable number of modes, with an emphasis on high quality for lower mode counts; this allows the synthesis algorithm to scale to different computational environments. A hybrid FIR/modal reverb architecture is also presented which allows for the efficient modeling of room impulse responses that contain sparse early reflections and dense late reverb. MUSHRA tests comparing the analysis/synthesis using various mode numbers for our algorithms, and for another state of the art algorithm, are included as well.
Download Surround Sound without Rear Loudspeakers: Multichannel Compensated Amplitude Panning and Ambisonics Conventional panning approaches for surround sound require loudspeakers to be distributed over the regions where images are needed. However in many listening situations it is not practical or desirable to place loudspeakers some positions, such as behind or above the listener. Compensated Amplitude Panning (CAP) is a method that adapts dynamically to the listener’s head orientation to provide images in any direction, in the frequency range up to ⇡ 1000 Hz using only 2 loudspeakers. CAP is extended here for more loudspeakers, which removes some limitations and provides additional benefits. The new CAP method is also compared with an Ambisonics approach that is adapted for surround sound without rear loudspeakers.
Download Fast Partial Tracking of Audio with Real-Time Capability through Linear Programming This paper proposes a new partial tracking method, based on linear programming, that can run in real-time, is simple to implement, and performs well in difficult tracking situations by considering spurious peaks, crossing partials, and a non-stationary shortterm sinusoidal model. Complex constant parameters of a generalized short-term signal model are explicitly estimated to inform peak matching decisions. Peak matching is formulated as a variation of the linear assignment problem. Combinatorially optimal peak-to-peak assignments are found in polynomial time using the Hungarian algorithm. Results show that the proposed method creates high-quality representations of monophonic and polyphonic sounds.