Download Audio Processing Systems
This tutorial1 gives an introduction into several hardware aspects for designing audio processing systems based on digital signal processors (DSP). Digital signal processors of different manufacturers and their use in practical circuits will be discussed.
Download Extracting Sinusoids From Harmonic Audio 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 with a digital signal processor.
Download Modulation And Delay Line Based Digital Audio Effects
In the field of musicians and recording engineers audio effects are mainly described and indicated by their acoustical effect. Audio effects can also be categorized from a technical point of view. The main criterion is found to be the type of modulation technique used to achieve the effect. After a short introduction to the different modulation types, three more sophisticated audio effect applications are presented, namely single sideband domain vibrato (mechanical vibrato bar simulation), a rotary speaker simulation, and an enhanced pitch transposing scheme.
Download Discrete-time Models for Non-linear Audio Systems
A variety of computational models have been proposed for digital simulation of nonlinear systems with memory [1, 2, 3, 4]. They are dealing with different aspects of the problem, like methods for identification, avoiding aliasing and fast convolution algorithms. In this paper we shortly sum up some of the common approaches and present a straightforward method for bandlimited discrete-time realization of analog nonlinear audio effects, like tube amps, exciters etc., using off-time digital cross correlation measurements. From these measurements we obtain a rather inefficient Wiener representation of the unknown nonlinearity. We then reduce the number of required coefficients significantly on the basis of multi-dimensional Laguerre transformation of the related Volterra kernels to allow real-time implementation on a digital signal processor [5].
Download Efficient linear prediction for digital audio effects
In many audio applications an appropriate spectral estimation from a signal sequence is required. A common approach for this task is the linear prediction [1] where the signal spectrum is modelled by an all-pole (purely recursive) IIR (infinite impulse response) filter. Linear prediction is commonly used for coding of audio signals leading to linear predictive coding (LPC). But also some audio effects can be created using the spectral estimation of LPC. In this paper we consider the use of LPC in a real-time system. We investigate several methods of calculating the prediction coefficients to have an almost fixed workload each sample. We present modifications of the autocorrelation method and of the Burg algorithm for a sample-based calculation of the filter coefficients as alternative for the gradient adaptive lattice (GAL) method. We discuss the obtained prediction gain when using these methods regarding the required complexity each sample. The desired constant workload leads to a fast update of the spectral model which is of great benefit for both coding and audio effects.
Download A Measurement Technique for Highly Nonlinear Transfer Functions
This paper presents a new method to estimate nonlinear transfer functions of tube amplifiers or distortion effect stages. A special test signal and a sorting algorithm allow the calculation of the nonlinear transfer functions. PSPICE simulations of a tube amplifier as well as real-time measurements of a tube amplifier with a high quality 24bit/96kHz sound card will be presented.
Download Performance Analysis of a Source Separation Algorithm
Source separation is an attractive preprocessing step for applying digital audio effects to a single source inside a signal mix. We present a performance analysis of a source separation algorithm based on time-frequency processing and its application to digital audio effects. The performance analysis gives insight to the main analysis parameters for the detection of the number of source signals inside the signal mix. We also analyze the main design parameters for the demixing operation which extracts a single source out of the signal mix.
Download Real-time implementation of a source separation algorithm
Source separation out of a mix of signals has been under development for many years with different approaches. We use timefrequency representations of two microphone signals to estimate the mixing parameters of the source signals. In order to evaluate the robustness of the algorithm under real-world conditions we built a real-time implementation, which is suitable to detect the sources, their mixing parameters and performs the source separation based on the mixing parameters. Our implementation only needs a few parameters and then works as a stand-alone solution with the opportunity to apply further post-processing or digital audio effects to the source signals.
Download Analysis of transient musical sounds by auto-regressive modeling
This paper gives an example of an auto-regressive (AR) spectral analysis on transient musical sounds. The attack part of many musical sounds is mostly too short to be analysed by a short-time Fourier analysis, whereas this short period of time is long enough for several AR-methods. The AR-spectra obtained from short segments of signals with attack transients have a sufficiently high frequency resolution. These spectra contain more information about the evolution of a sound than a fast Fourier transform made over a small amount of samples.
Download Digital Emulation of Analog Companding Algorithms for FM-Radio Transmission
Analog compander systems have been used to suppress the perception of noise in low dynamic range analog signal storage (tape recording) and signal transmission (FM radio). Commercial compander systems have been analyzed with respect to their signal processing requirements. The general structures of single- and multiband compander systems have been implemented on a high performance audio PC workstation. Audio tests and measurements with the optimized compander algorithms and parameters show very good performance. Even for transmission channels with very low signal-to-noise ratio (SNR of only 40 dB) an optimized digital multi-band compander emulation removes the channel noise perceptively from the output signal of the transmission system.