In this paper, a novel chroma extraction technique called TimeDomain Chroma Extraction (TDCE) is introduced. In comparison to many other known schemes, the calculation of a time-frequency representation is unnecessary since the TDCE is a pure sample-bysample technique. It mainly consists of a pitch tracking module that is implemented with a phase-locked loop (PLL). A set of 24 bandpass filters over two octaves is designed with the F 0 output of the pitch tracker to estimate a chroma vector. To verify the performance of the TDCE, a simple chord recognition algorithm is applied to the chroma output. The experimental results show that this novel time-domain chroma extraction technique yields good results while requiring only minor complexity and thus, enables the extraction of musical features in real-time on low-cost DSP platforms.

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In this study, receiver-based audio error concealment in the context of low-latency Audio over IP transmission is analyzed. Therefore, the well-known technique of audio extrapolation is investigated concerning its usability in real-time scenarios, its applied prediction techniques and various transmission parameters. A large-scale automated evaluation with PEAQ and a MUSHRA listening test reveal the performance of the various extrapolation setups. The results show the suitability of extrapolation to perform audio error concealment in real-time and the qualitative superiority of block based methods over sample based methods.

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We consider the problem of transmission errors in the well known adaptive differential pulse code modulation (ADPCM) system. A single transmission error destabilizes the reconstruction process at the decoder side in the ADPCM coding scheme if a non-leaky algorithm is used. We propose a delay-free and fixed rate of 3 bit/sample audio source coding scheme based on a robust prediction. The prediction of the backward ADPCM coding scheme is attained as a FIR filter in lattice structure. The prediction filter is derived as a reconstructed-signal-driven (RSD) or a predictionerror-driven (PED) lattice filter. A technique for an error robust RSD prediction is presented. This technique is employed in a robust audio coding scheme without use of any additional overhead. The proposed modified RSD-ADPCM is compared to the PED-ADPCM coding scheme by means of the objective audio quality. The proposed system yields good objective audio quality in the noise-free channels and provides robustness in the presence of transmission errors.

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In this study, a famous boxed effect pedal, also called stompbox, for electrical guitars is analyzed and simulated. The nodal DK method is used to create a non-linear state-space system with Matlab as a physical model for the MXR Phase 90 guitar effect pedal. A crucial component of the effect are Junction Field Effect Transistors (JFETs) which are used as variable resistors to dynamically vary the phase-shift characteristic of an allpass-filter cascade. So far, virtual analog modeling in the context of audio has mainly been applied to diode-clippers and vacuum tube circuits. This work shows an efficient way of describing the nonlinear behavior of JFETs, which are wide-spread in audio devices. To demonstrate the applicability of the proposed physical model, a real-time VST audio plug-in was implemented.

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In 1998, the ITU published a recommendation for an algorithm for objective measurement of audio quality, aiming to predict the outcome of listening tests. Despite the age, today only one implementation of that algorithm meeting the conformance requirements exists. Additionally, two open source implementations of the basic version of the algorithm are available which, however, do not meet the conformance requirements. In this paper, yet another non-conforming open source implementation, GstPEAQ, is presented. However, it improves upon the previous ones by coming closer to conformance and being computationally more efficient. Furthermore, it implements not only the basic, but also the advanced version of the algorithm. As is also shown, despite the nonconformance, the results obtained computationally still closely resemble those of listening tests.

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The sound of a vacuum tube guitar amplifier may be significantly influenced by the non-linear behavior of its output transformer, which therefore should also be considered in digital simulations. In this work, we develop a model for inductors and transformers with the magnetization following the model of Jiles and Atherton. For this purpose, the original magnetization model is rewritten to a differential equation with respect to time which can then easily be integrated into a previously developed circuit simulation framework. The model thus derived is then exercised in the simulation of three simple circuits where it shows the expected behavior.

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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.

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In the digital simulation of non-linear audio effect circuits, the arising non-linear equation system generally poses the main challenge for a computationally cheap implementation. As the computational complexity grows super-linearly with the number of equations, it is beneficial to decompose the equation system into several smaller systems, if possible. In this paper we therefore develop an approach to determine such a decomposition automatically. We limit ourselves to cases where an exact decomposition is possible, however, and do not consider approximate decompositions.

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The Ebers-Moll model has been widely used to represent Bipolar Junction Transistors (BJTs) in Virtual Analogue (VA) circuits. An investigation into the validity of this model is presented in which the Ebers-Moll model is compared to BJT models of higher complexity, introducing the Gummel-Poon model to the VA field. A comparison is performed using two complementary approaches: on fit to measurements taken directly from BJTs, and on application to physical circuit models. Targeted parameter extraction strategies are proposed for each model. There are two case studies, both famous vintage guitar effects featuring germanium BJTs. Results demonstrate the effects of incorporating additional complexity into the component model, weighing the trade-off between differences in the output and computational cost.

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Bucket brigade devices (BBDs) were invented in the late 1960s as a method of introducing a time-delay into an analog electrical circuit. They work by sampling the input signal at a certain clock rate and shifting it through a chain of capacitors to obtain the delay. BBD chips have been used to build a large variety of analog effects processing devices, ranging from chorus to flanging to echo effects. They have therefore attracted interest in virtual analog modeling and a number of approaches to modeling them digitally have appeared. In this paper, we propose a new model for the bucket-brigade device. This model is based on a variable samplerate, and utilizes the surrounding filtering circuitry found in real devices to avoid the need for the interpolation usually needed in such a variable sample-rate system.

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