Download Separating Piano Recordings into Note Events Using a Parametric Imitation Approach
In this paper we present a working system for separating a piano recording into events representing individual piano notes. Each note is parameterized with a transient-plus-harmonics model that, should all the parameters be reliably estimated, would produce near perfect reconstruction for each note as well as for the whole recording. However, interference between overlapping notes makes it hard to estimate parameters from their combination. In this work we propose to assess the estimability of sinusoidal parameters via their apparent degree of interference, estimate the estimable ones using algorithms suitable for different interference situations, and infer the hard-to-estimate parameters from the estimated ones. The outcome is a sequence of separate, parameterized piano notes that perceptually highly resemble, if are not identical to, the notes in the original recording. This allows for later analysis and processing stages using algorithms designed for separate notes.
Download Assessing The Suitability of the Magnitude Slope Deviation Detection Criterion For Use In Automatic Acoustic Feedback Control
Acoustic feedback is a recurrent problem in live sound reinforcement scenarios. Many attempts have been made to produce an automated feedback cancellation system, but none have seen widespread use due to concerns over the accuracy and transparency of feedback howl cancellation. This paper investigates the use of the Magnitude Slope Deviation (MSD) algorithm to intelligently identify feedback howl in live sound scenarios. A new variation on this algorithm is developed, tested, and shown to be much more computationally efficient without compromising detection accuracy. The effect of varying the length of the frequency spectrum history buffer available for analysis is evaluated across various live sound scenarios. The MSD algorithm is shown to be very accurate in detecting howl frequencies amongst the speech and classical music stimuli tested here, but inaccurate in the rock music scenario even when a long history buffer is used. Finally, a new algorithm for setting the depth of howl-cancelling notch filters is proposed and investigated. The algorithm shows promise in keeping frequency attenuation to a minimum required level, but the approach has some problems in terms of time taken to cancel howl.
Download A Robust Stochastic Approximation Method for Crosstalk Cancellation
Crosstalk cancellation serves as an important role in binaural signals playback through loudspeakers, which reproduce a particular auditory scene to the listener’s ears. In practice, due to either the listener’s head movement or rotation, etc, the actual transfer function matrix will differ from the design matrix, which results in deterioration in the performance of crosstalk cancellation. Crosstalk cancellation system (CCS) is very non-robust to these perturbations. Generally, in order to improve the robustness of CCS, several pairs of loudspeakers using a multi-band approach processing band-passed content to appropriately spaced loudspeakers are needed. In this paper, by means of assumed stochastic analysis, a stochastic robust approximation method based on random perturbation matrix modeling the variations of the transfer function matrix is introduced and evaluated. Under free-field condition, simulation results demonstrate the effectiveness of the proposed method.
Download Simulation of Analog Flanger Effect Using BBD Circuit
This paper deals with simulation of BBD circuit based analog flanger effects. The famous Electro-Harmonix Deluxe Electric Mistress flanger effect was used as a case study in this paper. The main attention of this paper is paid to the analysis and simulation of the LFO circuit, the BBD clock generator circuit and BBD circuit simulation of this effect. However, in order to compare the simulation results with measured data, the signal path simulation using the DK-method has been introduced as well.
Download Audio Nonlinear Modeling through Hyperbolic Tangent Functionals
In the present contribution we present the preliminary results of a black box nonlinear system (NLS) modeling. The NLS is composed by a nonlinear sigmoid-type input-output relationship (NLTF) followed by a linear system (LTI), as in a Hammerstein nonlinear system. Here, the used NLTF is derived from a deformation of the Hyperbolic Tangent power expansion. The advantage of using the hyperbolic tangent function is that nonlinearity depends on the linear and cubic terms that measure curvature (and thus nonlinearity) of the transfer function. The hyperbolic tangent model is extended to other types of nonlinear systems by expanding the nonlinear system in linear and increasingly nonlinear contributions, where the expansion parameters are deformed to enhance or suppress specific nonlinear modes of the expansion. Simulations were performed using Matlab 2012a. The preliminary results show fairly good agreement between the system obtained by parametric inference and a reference system, with mean square error (MSE)=0.035.
Download Signal-Matched Power-Complementary Cross-Fading and Dry-Wet Mixing
The blending of audio signals, called cross-fading, is a very common task in audio signal processing. Therefore, digital audio workstations offer several fading curves to select from. The choice of the fading curve typically depends on the signal characteristics and is supposed to result in a mixed signal featuring power and loudness close to the input signals. This work derives a correlationbased design of the fading curves to achieve exact power consistency to avoid audible fluctuations of the signal’s loudness. This principle is extended to the problem of mixing original signals with effect-processed signals using the dry-wet balance. Weighting coefficients for dry and wet signals are derived which realize the desired dry-wet balance but maintain the signal power.
Download Rounding Corners with BLAMP
The use of the bandlimited ramp (BLAMP) function as an antialiasing tool for audio signals with sharp corners is presented. Discontinuities in the waveform of a signal or its derivatives require infinite bandwidth and are major sources of aliasing in the digital domain. A polynomial correction function is modeled after the ideal BLAMP function. This correction function can be used to treat aliasing caused by sharp edges or corners which translate into discontinuities in the first derivative of a signal. Four examples of cases where these discontinuities appear are discussed: synthesis of triangular waveforms, hard clipping, and half-wave and fullwave rectification. Results obtained show that the BLAMP function is a more efficient tool for alias reduction than oversampling. The polynomial BLAMP can reduce the level of aliasing components by up to 50 dB and improve the overall signal-to-noise ratio by about 20 dB. The proposed method can be incorporated into virtual analog models of musical systems.
Download Time-Domain Implementation of a Stereo to Surround Sound Upmix Algorithm
This paper describes a time-domain algorithm to upmix stereo recordings for an enhanced playback on a surround sound loudspeaker setup. It is mainly the simplified version of a previously published frequency-domain algorithm where the standard shorttime Fourier transform is now replaced by an IIR filter bank. The design of complementary filter blocks and their arrangement in a tree structure to form a filter bank are derived. The arithmetic complexity of the filter bank itself and of the complete upmix algorithm is analysed and compared to the frequency-domain approach. The time-domain upmix is less flexible in its configuration but achieves an audio quality comparable to the frequency-domain implementation at a fraction of its computational cost.
Download Synthesis of Sound Textures with Tonal Components Using Summary Statistics and All-Pole Residual Modeling
The synthesis of sound textures, such as flowing water, crackling fire, an applauding crowd, is impeded by the lack of a quantitative definition. McDermott and Simoncelli proposed a perceptual source-filter model using summary statistics to create compelling synthesis results for non-tonal sound textures. However, the proposed method does not work well with tonal components. Comparing the residuals of tonal sound textures and non-tonal sound textures, we show the importance of residual modeling. We then propose a method using auto regressive modeling to reduce the amount of data needed for resynthesis and delineate a modified method for analyzing and synthesizing both tonal and non-tonal sound textures. Through user evaluation, we find that modeling the residuals increases the realism of tonal sound textures. The results suggest that the spectral content of the residuals has an important role in sound texture synthesis, filling the gap between filtered noise and sound textures as defined by McDermott and Simoncelli. Our proposed method opens possibilities of applying sound texture analysis to musical sounds such as rapidly bowed violins.
Download Reducing the Aliasing of Nonlinear Waveshaping Using Continuous-Time Convolution
Nonlinear waveshaping is a common technique in musical signal processing, both in a static memoryless context and within feedback systems. Such waveshaping is usually applied directly to a sampled signal, generating harmonics that exceed the Nyquist frequency and cause aliasing distortion. This problem is traditionally tackled by oversampling the system. In this paper, we present a novel method for reducing this aliasing by constructing a continuous-time approximation of the discrete-time signal, applying the nonlinearity to it, and filtering in continuous-time using analytically applied convolution. The presented technique markedly reduces aliasing distortion, especially in combination with low order oversampling. The approach is also extended to allow it to be used within a feedback system.