Download An Acoustic Paintbrush Method for Simulated Spatial Room Impulse Responses Virtual reality applications require all kinds of methods to create
plausible virtual acoustics environments to enhance the user experience. Here, we present an acoustic paintbrush method that modifies
the timbre of a simple room acoustics simulation with the timbre
of a measured room response while aiming to preserve the spatial
aspects of the simulated room. In other words, the method only
applies the measured spectral coloration and alters the simulated
and temporal distribution of early reflections as little as possible.
Three variations of the acoustic paintbrush method are validated
with a listening test. The results indicate that the method works
reasonably well. The paintbrushed room acoustic simulations were
perceived to become closer to the measured room acoustics than
the source simulation. However, the limits of the perceived effect
varied depending on the input signal and the simulated and recorded
responses. This warrants for further perceptual testing.
Download An Adaptive Technique For Modeling Audio Signals In many applications of audio signal processing modeling of the signal is required. The most commonly used approach for audio signal modeling is to assume the audio signal as an (autoregressive) AR-process where the audio signal is locally stationary over a relatively short time interval. In this case the audio signal can be modeled with an all-pole IIR (infinite impulse response) filter, which leads to LPC (linear predictive coding) where the current input sample is predicted by a linear combination of past samples of the input signal. However, in practice the relatively short time interval (i.e. a frame) where the signal is stationary will vary significantly in the audio signal data stream. Also the information content of the frames will show considerable variation. For a proper modeling of an audio signal it is essential that a suitable frame size and appropriate number of model parameters is used instead of a constant frame size and model order. In this paper we present an adaptive frame-by-frame technique for modeling audio signals, which automatically adjusts the optimal modeling frame size and the optimal number of model parameters for each frame.
Download Naturalness of Double-Slope Decay in Generalised Active Acoustic Enhancement Systems Active acoustic enhancement systems (AAESs) alter the perceived acoustics of a space by using microphones and loudspeakers to introduce sound energy into the room. Double-sloped energy decay may be observed in these systems. However, it is unclear as to which conditions lead to this effect, and to what extent double sloping reduces the perceived naturalness of the reverberation compared to Sabine decay. This paper uses simulated combinations of AAES parameters to identify which cases affect the objective curvature of the energy decay. A subjective test with trained listeners assessed the naturalness of these conditions. Using an AAES model, room impulse responses were generated for varying room dimensions, absorption coefficients, channel counts, system loop gains and reverberation times (RTs) of the artificial reverberator. The objective double sloping was strongly correlated to the ratio between the reverberator and passive room RTs, but parameters such as absorption and room size did not have a profound effect on curvature. It was found that double sloping significantly reduced the perceived naturalness of the reverberation, especially when the reverberator RT was greater than two times that of the passive room. Double sloping had more effect on the naturalness ratings when subjects listened to a more absorptive passive room, and also when using speech rather than transient stimuli. Lowering the loop gain by 9 dB increased the naturalness of the doublesloped stimuli, where some were rated as significantly more natural than the Sabine decay stimuli from the passive room.
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 MultiBin: A Binaural Audition Tool MultiBin is a new tool for binaural audition of multiple sound sources in a user definable environment. Although designed to be flexible in its application, its primary function is to provide dynamic multi-channel binaural simulation. It is built upon two new Csound binaural reverberation opcodes. An early reflection opcode, based on an image source method and a Head-Related Transfer Function interpolation algorithm previously introduced by the authors provides dynamic source and listener location. This is complemented by a later reverberation opcode which provides a diffuse reverb based on a parametric Feedback Delay Network model which considers interaural coherence.
Download Separation of Musical Instruments based on Perceptual and Statistical Principles The separation of musical instruments acoustically mixed in one source is a very active field which has been approached from many different viewpoints. This article compares the blind source separation perspective and oscillatory correlation theory taking the auditory scene analysis as the point of departure (ASA). The former technique deals with the separation of a particular signal from a mixture with many others from a statistical point of view. Through the standard Independent Component Analysis (ICA), a blind source separation can be done using the particular and the mixed signals' statistical properties. Thus, the technique is general and does not use previous knowledge about musical instruments. In the second approach, an ASA extension is studied with a dynamic neural model which is able to separate the different musical instruments taking a priori unknown perceptual elements as a point of departure. Applying an inverse transformation to the output of the model, the different contributions to the mixture can be recovered again in the time domain.
Download Bunkervik Spatial Reverb Demo This paper accompanies a demonstration of a real-time audio plug-in for a dynamic spatial reverb. The reverb is based on acoustic measurements of the Bunkervik creative arts space in Brescia, Italy. A modal synthesis reverberation engine was created based on measured impulse responses from three locations in the tunnel. Using a common set of modal frequencies the position of the receiver can be dynamically moved through the space by interpolating between data sets of residue weights and FIR filter taps. The audio plug-in also allows real-time manipulation of the high-frequency content, damping, and microphone rotation, all of which can be modulated using two LFOs.
Download Faust2android: a Faust Architecture For Android faust2android is a tool that turns a FAUST program into an Android application. Signal processing tasks as well as accessing the audio record and playback resources are done natively in C++ using the Android Native Development Toolkit (NDK). User interface and other components of the application are programmed in JAVA. The implementation as well as issues related to real-time signal processing on Android platforms are discussed. faust2android is part of a larger project whose goal is to build a full FAUST environment for Android: FAUST D ROID.
Download SMSPerformer: A real-time synthesis interface for SMS SmsPerformer is a graphical interface for the real-time SMS synthesis engine. The application works from analyzed sounds and it has been designed to be used both as a composition and a performance tool. The program includes programmable time-varying transformations, MIDI control for the synthesis parameters, and performance loading and saving options.
Download Non-Linear Digital Implementation of a Parametric Analog Tube Ground Cathode Amplifier In this paper we propose a digital simulation of an analog amplifier circuit based on a grounded-cathode amplifier with parametric tube model. The time-domain solution enables the online valve model substitution and zero-latency changes in polarization parameters. The implementation also allows the user to match various types of tube processing features.