Download Harmonizing effect using short-time time-reversal
A prior study of short-time time-reversal showed sideband modulation occurs for short time durations, creating overtones for single sinusoid signals. In this paper, we examine the overtones created by short-time time-reversal and the tonal relation between the overtones and the input signal. We present three methods of using short-time time-reversal for harmonizing audio signals. Then modifications to the previous short-time time-reversal needed to implement the proposed methods are described.
Download Barberpole Phasing and Flanging Illusions
Various ways to implement infinitely rising or falling spectral notches, also known as the barberpole phaser and flanging illusions, are described and studied. The first method is inspired by the Shepard-Risset illusion, and is based on a series of several cascaded notch filters moving in frequency one octave apart from each other. The second method, called a synchronized dual flanger, realizes the desired effect in an innovative and economic way using two cascaded time-varying comb filters and cross-fading between them. The third method is based on the use of single-sideband modulation, also known as frequency shifting. The proposed techniques effectively reproduce the illusion of endlessly moving spectral notches, particularly at slow modulation speeds and for input signals with a rich frequency spectrum. These effects can be programmed in real time and implemented as part of a digital audio processing system.
Download Distortion and Pitch Processing Using a Modal Reverberator Architecture
A reverberator based on a room response modal analysis is adapted to produce distortion, pitch and time manipulation effects, as well as gated and iterated reverberation. The so-called “modal reverberator” is a parallel collection of resonant filters, with resonance frequencies and dampings tuned to the modal frequencies and decay times of the space or object being simulated. Here, the resonant filters are implemented as cascades of heterodyning, smoothing, and modulation steps, forming a type of analysis/synthesis architecture. By applying memoryless nonlinearities to the modulating sinusoids, distortion effects are produced, including distortion without intermodulation products. By using different frequencies for the heterodyning and associated modulation operations, pitch manipulation effects are generated, including pitch shifting and spectral “inversion.” By resampling the smoothing filter output, the signal time axis is stretched without introducing pitch changes. As these effects are integrated into a reverberator architecture, reverberation controls such as decay time can be used produce novel effects having some of the sonic characteristics of reverberation.
Download Stereo signal separation and upmixing by mid-side decomposition in the frequency-domain
An algorithm to estimate the perceived azimuth directions in a stereo signal is derived from a typical signal model. These estimated directions can then be used to separate direct and ambient signal components and to remix the original stereo track. The processing is based on the idea of a bandwise mid-side decomposition in the frequency-domain which allows an intuitive and easy to understand mathematical derivation. An implementation as a stereo to five channel upmix is able to deliver a high quality surround experience at low computational costs and demonstrates the practical applicability of the presented approach.
Download Automatic subgrouping of multitrack audio
Subgrouping is a mixing technique where the outputs of a subset of audio tracks in a multitrack are summed to a single audio bus. This is done so that the mix engineer can apply signal processing to an entire subgroup, speed up the mix work flow and manipulate a number of audio tracks at once. In this work, we investigate which audio features from a set of 159 can be used to automatically subgroup multitrack audio. We determine a subset of audio features from the original 159 audio features to use for automatic subgrouping, by performing feature selection using a Random Forest classifier on a dataset of 54 individual multitracks. We show that by using agglomerative clustering on 5 test multitracks, the entire set of audio features incorrectly clusters 35.08% of the audio tracks, while the subset of audio features incorrectly clusters only 7.89% of the audio tracks. Furthermore, we also show that using the entire set of audio features, ten incorrect subgroups are created. However, when using the subset of audio features, only five incorrect subgroups are created. This indicates that our reduced set of audio features provides a significant increase in classification accuracy for the creation of subgroups automatically.
Download Separation of musical notes with highly overlapping partials using phase and temporal constrained complex matric factorization
In note separation of polyphonic music, how to separate the overlapping partials is an important and difficult problem. Fifths and octaves, as the most challenging ones, are, however, usually seen in many cases. Non-negative matrix factorization (NMF) employs the constraints of energy and harmonic ratio to tackle this problem. Recently, complex matrix factorization (CMF) is proposed by combining the phase information in source separation problem. However, temporal magnitude modulation is still serious in the situation of fifths and octaves, when CMF is applied. In this work, we investigate the temporal smoothness model based on CMF approach. The temporal ac-tivation coefficient of a preceding note is constrained when the succeeding notes appear. Compare to the unconstraint CMF, the magnitude modulation are greatly reduced in our computer simulation. Performance indices including sourceto-interference ratio (SIR), source-to-artifacts ratio (SAR), sourceto-distortion ratio (SDR), as well as modulation error ratio (MER) are given.
Download Automatic calibration and equalization of a line array system
This paper presents an automated Public Address processing unit, using delay and magnitude response adjustment. The aim is to achieve a flat frequency response and delay adjustment between different physically-placed speakers at the measuring point, which is nowadays usually made manually by the sound technician. The adjustment is obtained using three signal processing operations to the audio signal: time delay adjustment, crossover filtering, and graphic equalization. The automation is in the calculation of different parameter sets: estimation of the time delay, the selection of a suitable crossover frequency, and calculation of the gains for a third-octave graphic equalizer. These automatic methods reduce time and effort in the calibration of line-array PA systems, since only three sine sweeps must be played through the sound system. Measurements have been conducted in an anechoic chamber using a 1:10 scale model of a line array system to verify the functioning of the automatic calibration and equalization methods.
Download AM/FM DAFx
In this work we explore audio effects based on the manipulation of estimated AM/FM decomposition of input signals, followed by resynthesis. The framework is based on an incoherent monocomponent based decomposition. Contrary to reports that discourage the usage of this simple scenario, our results have shown that the artefacts introduced in the audio produced are acceptable and not even noticeable in some cases. Useful and musically interesting effects were obtained in this study, illustrated with audio samples that accompany the text. We also make available Octave code for future experiments and new Csound opcodes for real-time implementations.
Download On comparison of phase alignments of harmonic components
This paper provides a method for comparing phase angles of harmonic sound sources. In particular, we propose an algorithm for decomposing the difference between two sets of phases into a harmonic part, which represents the phase progress of harmonic components, and a residue part, which represents all causes of deviations from perfect harmonicity. This decomposition allows us to compare phase alignments regardless of an arbitrary time shift, and handle harmonic and noise/inharmonic parts of the phase angle separately to improve existing algorithms that handles harmonic sound sources using phase measurements. These benefits are demonstrated with a new phase-based pitch marking algorithm and an improved time-scale and pitch modification scheme using traditional harmonic sinusoidal modelling.
Download Towards Transient Restoration in Score-informed Audio Decomposition
Our goal is to improve the perceptual quality of transient signal components extracted in the context of music source separation. Many state-of-the-art techniques are based on applying a suitable decomposition to the magnitude of the Short-Time Fourier Transform (STFT) of the mixture signal. The phase information required for the reconstruction of individual component signals is usually taken from the mixture, resulting in a complex-valued, modified STFT (MSTFT). There are different methods for reconstructing a time-domain signal whose STFT approximates the target MSTFT. Due to phase inconsistencies, these reconstructed signals are likely to contain artifacts such as pre-echos preceding transient components. In this paper, we propose a simple, yet effective extension of the iterative signal reconstruction procedure by Griffin and Lim to remedy this problem. In a first experiment, under laboratory conditions, we show that our method considerably attenuates pre-echos while still showing similar convergence properties as the original approach. A second, more realistic experiment involving score-informed audio decomposition shows that the proposed method still yields improvements, although to a lesser extent, under non-idealized conditions.