Udo Zölzer

Adaptive digital audio effects (a-DAFx): a new class of sound transformations

After covering the basics of sound perception and giving an overview of commonly used audio effects (using a perceptual categorization), we propose a new concept called adaptive digital audio effects (A-DAFx). This consists of combining a sound transformation with an adaptive control. To create A-DAFx, low-level and perceptual features are extracted from the input signal, in order to derive the control values according to specific mapping functions. We detail the implementation of various new adaptive effects and give examples of their musical use

Extracting Sinusoids from Harmonic Signals

This paper presents a special window function for a Fast Fourier Transform (FFT) based spectral modeling approach for signals consisting of sinusoids plus noise. The main new idea is to choose a time window function with a simple Fourier transform. With the knowledge of the Fourier transform of the window function we are able to extract the parameters (frequency, amplitude, and phase) of sinusoids in real-time.

Introduction to the Special Issue on Virtual Analog Audio Effects and Musical Instruments

The 16 papers in this special issue focus on virtual audio effects and musical instruments.

Multi-complementary filter bank

An octave-spaced complementary filter bank that provides aliasing-free subband signal processing and perfect reconstruction is presented. The transition bandwidth of the splitting filters can be controlled independently by basis filters, which in turn are realized by multirate complementary filters. The design of the filter bank is presented.<<ETX>>

Audio signal decomposition for pitch and time scaling

This paper describes a frame based audio signal decomposition approach. The aim of the proposed approach is first to decompose an audio signal into transient, sinusoidal and residual components (TSR). Applying linear prediction onto each signal frame, transient detection is started on the prediction error. Assuming that sudden changes like transients are difficult to predict, the prediction error is thus expected to have very high energy in transient areas. Using the estimated envelope of the prediction error and its first order statistical moments, a suitable adaptive threshold is derived which ensures correct transient detection in various audio signals. Once a transient region is detected in a signal frame, it is directly separated yielding a first residual signal containing sinusoids and noise. To extract the sinusoidal components, partial tracking based on psychoacoustic masking is performed on the first residual signal. A second residual signal is then obtained by subtracting the psychoacoustic relevant sinusoidal components. This second residual is then processed by tracking and removing remaining sinusoidal components yielding a final residual without transient and sinusoidal components. Pitch and time scaling can then be applied onto the decomposed signal components. Pitch scaling is only applied onto the sinusoidal components, transient and residual components are not changed. Time scaling is applied here to the sinusoidal and residual components, whereas the transient components are only shifted in time.

New Approach for Resolving Ambiguities for Semi-Blind Equalization of MIMO Frequency Selective Channels

This paper addresses the semi-blind equalization for a wireless multiple-input multiple-output (MIMO) system with frequency selective channels. When reformulating the convolutive blind signal separation problem to allow the application of ordinary complex independent component analysis (ICA), the number of permutations is increased. An algorithm presented here reduces the ambiguities to those of the non-frequency selective case and is applicable for channels with an arbitrary power delay profile. The remaining ambiguities can be solved by a short preamble. The efficiency of the proposed method is illustrated by numerical simulations. According to the bit error rate the semi-blind equalization shows a good performance in comparison to training based channel estimation & equalization

Parametric second- and fourth-order shelving filters for audio applications

In this paper, we develop a parametric recursive second-order lowpass/highpass shelving filters and fourth-order bandpass shelving filters. These filters allow a better frequency selectivity for audio equalization as compared to the well-known first-order lowpass/highpass shelving filters and second-order peak filters. In the parametric structures, the cut-off frequencies and the gain factors can be set independently. We derive the filter designs for the discrete-time implementation. Furthermore, we apply a technique for the implementation of delay-free loops which is necessary to yield the exact inverse filter to realize symmetric boost and cut of a frequency band.

MIMO-OFDM on a flexible laboratory radio system

In this paper we present a flexible MIMO-OFDM system concept for research purposes. Algorithms for multiplexing pilot tones into MIMO-OFDM signals are reviewed. A general formulation for using orthogonal pilot sequences that extent across multiple OFDM symbols is given. Simulations are carried out to evaluate the performance of the algorithms. Transmissions are performed on a MIMO laboratory system in an indoor LOS (line of sight) environment with various parameters to validate the useability of the proposed concept.

Delay-free lossy audio coding using shelving pre- and post-filters

A delay-free audio coding scheme based on ADPCM with adaptive pre- and post-filtering is presented. The pre-/post- filters are realized as a cascade of shelving filters, designed to match the characteristics of human perception. The pre- and post-filters are adapted by dynamic compression of the respective sub-bands. The adaption is backward-adaptive, i.e. is fed by the reconstructed signal, which eliminates the need to transmit the filter coefficients and allows delay-free operation. This pre- and post-filtering significantly improves the audio quality compared to a plain ADPCM codec, as underlined by objective measurements. Since the base ADPCM used is also delay-free, the resulting coding system works without any algorithmic delay.

Ultra-low delay lossy audio coding using DPCM and block companded quantization

Real-time audio transmission requires good quality for a restricted channel capacity and minimum latency. An ultra-low delay audio coding scheme based on differential pulse code modulation (DPCM) and block companded quantization is presented. The prediction filter of the base backward DPCM codec is attained as a FIR filter in lattice structure. The proposed block based audio coding introduces an algorithmic latency below one millisecond and an overhead of less than a half bit per sample. The advantage of the block companded quantization with DPCM is the capability to follow rapid changes in the signal. Therefore, it significantly improves the perceptual audio quality compared to a plain DPCM coding scheme with an adaptive quantizer.