Download An auditorily motivated analysis method for room impulse responses In this paper a new auditorily motivated analysis method for room impulse responses is presented. The method applies same kind of time and frequency resolution than the human hearing. With the proposed method it is possible to study the decaying sound field of a room in more detail. It is applicable as well in the analysis of artificial reverberation and related audio effects. The method, used with directional microphones, gives us also hints about the diffuseness and the directional characteristics of the sound fields in the time-frequency domain. As a case study two example room impulse responses are analyzed.
Download Separation of overlapping impulsive sounds by bandwise noise interpolation The task of extracting harmonic content of multiple pitched sources from a mono audio mix has been investigated on several occasions [1, 2, 3, 4]. However, most pitched notes contain an inharmonic component, which is an important perceptual attribute of the sound. This content is usually not dealt with during separation. It would also be interesting in its own right to develop separation techniques for extracting percussive sounds for polyphonic mixes. This paper describes an attempt at separating overlapping impulsive content of multiple sources from a mono mix. The method uses an interpolation within individual frequency bands of the decaying noise envelope of each source across overlapping sections with other sources. Three analysis methods determining the distribution of these bands were tested: the DFT followed by processing in Bark bands, the discrete wavelet transform (DWT), and the dyadic wavelet packet transform (DWPT).
Download Music Dereverberation by Spectral Linear Prediction in Live Recordings In this paper, we present our evaluations in using blind single channel dereverberation on music signals. The target material is heavily reverberated and dynamic range compressed polyphonic music from several genres. The applied dereverberation method is based on spectral subtraction regulated by a time-frequency domain linear predictive model. We present our results on enhancing music signal quality and automatic beat tracking accuracy with the proposed dereverberation method. Signal quality enhancement, measured by improvement in signal to distortion ratio, is achieved for both reverberant and dynamic range compressed signals. Moreover, the algorithm shows potential as a preprocessing method for music beat tracking.
Download Fast Parametric Matrices for Lossless Feedback Delay Networks This paper presents a framework for designing creative reverbs using parametric orthogonal feedback matrices on Feedback Delay Networks (FDNs) through recursive Kronecker products of 2D rotation and reflection matrices. By parameterizing each 2×2 kernel with a single angle, we construct a family of 2M×2M orthogonal matrices that maintain losslessness while enabling continuous control over network topology. We then exploit their recursive definition to compute the feedback operation with an O(N log₂ N) divide-and-conquer algorithm that matches the Fast Walsh-Hadamard Transform time complexity while offering parametric flexibility. Strategic manipulation of individual kernel angles enables creative sound design applications, such as stereo cross-coupling, selective freeze, and time-varying modulation for resonance breaking.
Download Variable Pre-Emphasis LPC for Modeling Vocal Effort in the Singing Voice In speech and singing, the spectral envelope of the glottal source varies according to different voice qualities such as vocal effort, lax voice, and breathy voice. In contrast, linear prediction coding (LPC) models the glottal source in a way that is not flexible. The spectral envelope of the source estimated by LPC is fixed and determined by the pre-emphasis filter. In standard LPC, the formant filter captures variation in the spectral envelope that should be associated with the source. This paper presents variable preemphasis LPC (VPLPC) as a technique to allow the estimated source to vary. This results in formant filters that remain more consistent across variations in vocal effort and breathiness. VPLPC also provides a way to change the envelope of the estimated source, thereby changing the perception of vocal effort. The VPLPC algorithm is used to manipulate some voice excerpts with promising but mixed results. Possible improvements are suggested.
Download Gesturally-Controlled Digital Audio Effects This paper presents a detailed analysis of the acoustic effects of the movements of single-reed instrument performers for specific recording conditions. These effects are shown to be mostly resulting from the difference between the time of arrival of the direct sound and that of the first reflection, creating a sort of phasing or flanging effect. Contrary to the case of commercial flangers – where delay values are set by a LFO (low frequency oscillator) waveform – the amount of delay in a recording of an acoustic instrument is a function of the position of the instrument with respect to the microphone. We show that for standard recordings of a clarinet, continuous delay variations from 2 to 5 ms are possible, producing a naturally controlled effect.
Download CMOS Implementation of an Adaptive Noise Canceller into a Subband Filter In recent years the demand for mobile communication has increased rapidly. While in the early years of mobile phones battery life was one of the main concerns for developers speech quality is now becoming one of the most important factors in the development of the next generation of mobile phones. This paper describes the CMOS implementation of an adaptive noise canceller (ANC) into a subband filter. The ANC-Subband filter is able to reduce noise components of real speech without prior knowledge of the noise properties. It is predestined to be used in mobile devices and therefore, uses a very low clock frequency resulting in a small power consumption. This low power consumption combined with its small physical size enables the circuit also be used in hearing aids to efficiently reduce noise contained in the speech signal.
Download Using Ideas from Natural Selection to Evolve Synthesized Sounds This paper describes a system for the automatic creation of digital synthesizer circuits that can generate sounds similar to a sampled (target) sound. The circuits will consist of very basic signal functions and generators that are arbitrarily interconnected. The system uses a “genetic algorithm” (GA) to evolve successively better circuits. First it creates populations of such synthesizers, generates the output and a fitness value of each individual circuit. The ones that are best at imitating the target sound will be kept. They are used for “breeding” to form a new generation where, hopefully, at least some individuals perform better than their parents did. The end result will be a circuit that can create a sound that resembles the target sample. Because it’s a synthesizer we can manipulate the different parameters when generating the sound. We can also get a very compact representation of the sound that can be useful when distributing music over a limited bandwidth communications channel (e.g. Internet). As we shall see, it also gives the user a very powerful tool for creating totally new sounds.
Download Intermodulation Effects Analysis using Complex Bandpass Filterbanks The objective of this paper is to show the ability of complex bandpass filterbanks to extract the intermodulation information that appears when two audio signals interact inside the same analysis band. To perform the analysis a sinusoidal model of the signals has been assumed. Three kinds of signals have been analyzed: a sum of two cosines, a sum of two linear chirps and a sum of two exponential chirps. The complex bandpass filtering of the signals is carried out using a new algorithm based on the Complex Continuous Wavelet Transform. The developed algorithm has been validated comparing the practical results with the theoretical instantaneous amplitude and instantaneous phase of the obtained model of the signals. With the appropriate width, the complex bandpass filters show the same behaviour as our perceptual ability to discriminate interacting tones when they fall inside a critical band of the human ear.
Download Praat AudioTools: Analysis Objects as Compositional Controllers for Interpretable Sound Transformation This demonstration presents Praat AudioTools, an open-source hybrid toolkit that repurposes Praat's phonetic-analysis environment for electroacoustic composition, sound design, and offline analysis–resynthesis workflows. Rather than treating analysis data as temporary measurements hidden inside an audio processor, Praat AudioTools exposes pitch contours, formant structures, temporal segmentations, spectral descriptors, phrase boundaries, stochastic trajectories, and host-application exchange files as editable compositional objects. These objects can be inspected, modified, chained, reused, and rendered into new sound transformations. The demonstration focuses on seven offline workflows: Neural Ambient Drone Designer, Praat for Max and Max for Live, Phase-Space Composer, Reich Generator, MCMC Musical Variation, Messagesquisse Opening, and Vector/Full-Chain composition workflows. None of the examples are presented as real-time effects. Instead, they show an "edit-in-the-middle" model in which sound is analyzed, intermediate representations are made visible, compositional decisions are applied to those representations, and the result is rendered as audio. The aim is to demonstrate a transparent alternative to both conventional black-box audio effects and end-to-end generative audio systems: a compositional environment where analysis objects become controllers, traces, scores, and reproducible technical artifacts.