Download Spectral Dealy Filters with Feedback Delay Filters with Feedback and Time-Varying Coefficients A recently introduced structure to implement a continuously smooth spectral delay, based on a cascade of first-order allpass filters and an equalizing filter, is described and the properties of this spectral delay filter are reviewed. A new amplitude envelope equalizing filter for the spectral delay filter is proposed and the properties of structures utilizing feedback and/or time-varying filter coefficients are discussed. In addition, the stability conditions for the feedback and the time-varying structures are derived. A spectral delay filter can be used for synthesizing chirp-like sounds or for modifying the timbre of arbitrary audio signals. Sound examples on the use of the spectral delay filters utilizing the structures discussed in this paper can be found at http://www.acoustics.hut. fi/publications/papers/dafx09-sdf/.
Download Compositional Sketches in PWGLSynth PWGLSynth has already a long history in controlling physicsbased instruments. The control system has been score-based, i.e. the user prepares a score in advance, and by interactive listening process the result can be be refined either by adjusting score information, performance rules and/or the visual instrument definition. This scheme allows detailed control on how the instrument model reacts to control information generated from the score. This paper presents a complementary approach to sound synthesis where the idea is to generate algorithmically typically relatively short musical textures. The user can improvise with various compositional ideas, adjust parameters, and listen to the results in real-time either individually or interleaved. This is achieved by utilizing a special code-box scheme that allows any textual Lisp expression to be interfaced to the visual part of the PWGL system.
Download Sound synthesis using an allpass filter chain with audio‐rate coefficient modulation This paper describes a sound synthesis technique that modulates the coefficients of allpass filter chains using audio-rate frequencies. It was found that modulating a single allpass filter section produces a feedback AM–like spectrum, and that its bandwidth is extended and further processed by non-sinusoidal FM when the sections are cascaded. The cascade length parameter provides dynamic bandwidth control to prevent upper range aliasing artifacts, and the amount of spectral content within that band can be controlled using a modulation index parameter. The technique is capable of synthesizing rich and evolving timbres, including those resembling classic virtual analog waveforms. It can also be used as an audio effect with pitch-tracked input sources. Software and sound examples are available at http://www.acoustics.hut.fi/publications/papers/dafx09-cm/
Download The Influence of Small Variations in a Simplified Guitar Amplifier Model A strongly simplified guitar amplifier model, consisting of four stages, is presented. The exponential sweep technique is used to measure the frequency dependent harmonic spectra. The influence of small variations of the system parameters on the harmonic components is analyzed. The differences of the spectra are explained and visualized.
Download Trans-synthesis System for Polyphonic Musical Recordings of Bowed-String Instruments A system that tries to analyze polyphonic musical recordings of bowed-string instruments, extract synthesis parameters of individual instrument and then re-synthesize is proposed. In the analysis part, multiple F0s estimation and partials tracking are performed based on modified WGCDV (weighted greatest common divisor and vote) method and high-order HMM. Then, dynamic time warping algorithm is employed to align the above results with the score to improve the accuracy of the extracted parameters. In the re-synthesis part, simple additive synthesis is employed. Here, one can experiment on changing timbres, pitches and so on or adding vibrato or other effects on the same piece of music.
Download Audio FFT Filter Banks FFT-based nonuniform filter banks are proposed based on channelsized inverse FFTs applied to nonuniform frequency-partitions (or overlap-add decompositions) of the Short Time Fourier Transform (STFT). Audio filter banks (particularly octave filter banks) are considered as application examples. Trade-offs discussed include perfect reconstruction, aliasing cancellation, flexibility of filterchannel band edges, use of the FFT for speed, multirate timedomain channel signals, time-varying filtering, and associated issues.
Download KRONOS ‐ A Vectorizing Compiler for Music DSP This paper introduces Kronos, a vectorizing Just in Time compiler designed for musical programming systems. Its purpose is to translate abstract mathematical expressions into high performance computer code. Musical programming system design criteria are considered and a three-tier model of abstraction is presented. The low level expression Metalanguage used in Kronos is described, along with the design choices that facilitate powerful, yet transparent vectorization of the machine code.
Download Human Inspired Auditory Source Localization This paper describes an approach for the localization of a sound source in the complete azimuth plane of an auditory scene using a movable human dummy head. A new localization approach which assumes that the sources are positioned on a circle around the listener is introduced and performs better than standard approaches for humanoid source localization like the Woodworth formula and the Freefield formula. Furthermore a localization approach based on approximated HRTFs is introduced and evaluated. Iterative variants of the algorithms enhance the localization accuracy and resolve specific localization ambiguities. In this way a localization blur of approximately three degrees is achieved which is comparable to the human localization blur. A front-back confusion allows a reliable localization of the sources in the whole azimuth plane in up to 98.43 % of the cases.
Download Binaural HRTF-based Spatialization: New Approaches and Implementation New approaches to Head Related Transfer Function (HRTF) based artificial spatialisation of audio are presented and discussed in this paper. A brief summary of the topic of audio spatialisation and HRTF interpolation is offered, followed by an appraisal of the existing minimum phase HRTF interpolation method. Novel alternatives are then suggested which essentially approach the problem of phase interpolation more directly. The first technique, based on magnitude interpolation and phase truncation, aims to use the empirical HRTFs without the need for complex data preparation or manipulation, while minimizing any approximations that may be introduced by data transformations. A second approach augments a functionally based phase model with low frequency non-linear frequency scaling based on the empirical HRTFs, allowing a more accurate phase representation of the more relevant lower frequency end of the spectrum. This more complex approach is deconstructed from an implementation point of view. Testing of both algorithms is then presented, which highlights their success, and favorable performance over minimum phase plus delay methods.
Download A FPGA‐based Adaptive Noise Cancelling System A FPGA-based system suitable for augmented reality audio applications is presented. The sample application described here is adaptive noise cancellation (ANC). The system consists of a Spartan -3 FPGA XC3S400 board connected to a Philips Stereo-AudioCodec UCB 1400. The algorithms for the FIR filtering and for the adaption of the filter coefficients according to the Widrow-Hoff LMS algorithm are implemented on the FPGA board. Measurement results obtained with a dummy head measuring system are reported, and a detailed analysis of system performance and possible system improvements is given.