Martin Holters

Martin Holters
Helmut Schmidt University / University of the Federal Armed Forces Hamburg | HSU · Department of Signal Processing and Communication

Dr.-Ing.

About

68
Publications
59,434
Reads
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558
Citations
Additional affiliations
March 2009 - present
Helmut Schmidt University / University of the Federal Armed Forces Hamburg
Position
  • Professor (Assistant)
January 2005 - February 2009
Helmut Schmidt University / University of the Federal Armed Forces Hamburg
Position
  • Research Assistant

Publications

Publications (68)
Conference Paper
Full-text available
The main characteristic of shelving fiters, as commonly used in audio equalization, is to amplify or attenuate a certain frequency band by a given gain. For parametric equalizers, a fiter structure is desirable that allows independent adjustment of the width and center frequency of the band, and the gain. In this paper, we present a design for arbi...
Conference Paper
Full-text available
Real-time bidirectional audio applications, like microphones and monitor speakers in live performances, typically require communication systems with minimum latency. When digital transmission with limited bit rate is desired, this poses tight constraints on the algorithmic delay of the audio coding scheme. We present a delay-free approach employing...
Conference Paper
Full-text available
The nodal DK method is a systematic way to derive a non-linear state-space system as a physical model for an electrical circuit. Unfortunately, calculating the system coefficients requires inversion of a relatively large matrix. This becomes a problem when the system changes over time, requiring continuous recomputation of the coefficients. In this...
Conference Paper
Full-text available
Digital emulation of analog circuits for musical audio processing , like synthesizers, guitar effect pedals, or vintage amplifiers , is an ongoing research topic. David Yeh proposed to use the nodal DK method to derive a non-linear state-space system from a circuit schematic in a very systematic way. However, this approach has some drawbacks and li...
Article
Full-text available
Nonlinear systems, such as guitar distortion effects, play an important role in musicalsignal processing. One major problem encountered in digital nonlinear systems is aliasing distortion.Consequently, various aliasing reduction methods have been proposed in the literature. One ofthese is based on using the antiderivative of the nonlinearity and ha...
Conference Paper
Full-text available
The Red Llama guitar overdrive effect pedal differs from most other overdrive effects because it utilizes CMOS inverters, formed by two metal-oxide-semiconductor field-effect transistors (MOSFETs), instead of a combination of operational amplifiers and diodes to obtain nonlinear distortion. This makes it an interesting subject for virtual analog mo...
Conference Paper
Full-text available
Nonlinear systems, like e.g. guitar distortion effects, play an important role in musical signal processing. One major problem encountered in digital nonlinear systems is aliasing distortion. Consequently , various aliasing reduction methods have been proposed in the literature. One of these is based on using the antideriva-tive of the nonlinearity...
Conference Paper
Full-text available
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 devi...
Conference Paper
Full-text available
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...
Conference Paper
Full-text available
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 pos...
Conference Paper
Full-text available
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 loud-ness close to the...
Conference Paper
Full-text available
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 or...
Conference Paper
Full-text available
In the digital simulation of non-linear audio effect circuits, the arising non-linear equation generally poses the main challenge for a computationally cheap implementation. For any but the simplest circuits, using an iterative solver at execution time will be too slow, while exhaustive look-up tables quickly grow intolerably large. To better cope...
Conference Paper
Full-text available
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 a...
Data
Full-text available
Conference Paper
Full-text available
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 ef-fect pedal. A crucial component of the effect are Junction Field Effect Transistors (JFETs) which are...
Conference Paper
This work comprises an extension of a backward adaptive quantizer which is employed together with a robust lattice predictor in an ADPCM coding scheme. Predictors of the ADPCM audio coding schemes are often considered as the part most sensitive to transmission errors. Nevertheless, a single transmission error causes a short destabilization of the a...
Conference Paper
In this paper, a load balanced implementation of a delayless FxLMS algorithm for the purpose of active noise cancellation is proposed. Frequency-domain adaption algorithms using FFT’s are well-known for their efficiency. However, their block-based character will lead to a delay in the order of the used block length. A hybrid adaption approach is ch...
Conference Paper
Full-text available
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 \unit[3]{bit/sample} audio source...
Conference Paper
Full-text available
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 predic-tion techniques and various transmission parameters. A large-scale automated evalua...
Conference Paper
Full-text available
A low delay audio coding scheme with good perceptual au-dio quality for a desired limited bit rate is presented. The proposed audio coding scheme is based on differential pulse code modulation (DPCM) and block companded (BC) quanti-zation. Prediction is realized as a FIR filter in lattice structure. DPCM performs in feedback manner, therefore no tr...
Conference Paper
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...
Conference Paper
High Frequency Surface Wave Radars (HFSWRs) play an important role in long-range ocean surveillance, with particular interest in reliable detection and tracking of far-distant ships. One of the biggest challenges in ship target detection in HFSWR is the non-homogeneous detection background. Depending on the chosen detection parameters, this non-hom...
Conference Paper
Full-text available
Virtual analog synthesis requires bandlimited source signal algorithms. An efficient methodology for the task expresses the traditionally used source waveforms or their time-derivatives as a sequence of bandlimited impulses or step functions. Approximations of the ideal bandlimited functions used in these quasi-bandlimited oscillator algorithms are...
Conference Paper
Full-text available
This paper deals with usage of approximations for simulation of more complex audio circuits. A Fender type guitar preamp was chosen as a case study. This circuit contains two tubes and thus four nonlinear functions as well as it is a parametric circuit because of an integrated tone stack. A state-space approach was used for simulation and further,...
Conference Paper
Full-text available
In this paper, a novel chroma extraction technique called Time-Domain 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-...
Conference Paper
Full-text available
An advanced phase vocoder technique for high quality audio pitch shifting and time stretching is described. Its main concept is based on the PVSOLA time stretching algorithm which is already known to give good results on monophonic speech. Some enhancements are proposed to add the ability to process polyphonic material at equal quality by distingui...
Conference Paper
Common tasks of High-Frequency Surface Wave Radars (HFSWRs) are long-range ocean state monitoring and maritime surveillance with a strong focus on the detection of ships. Due to the heterogeneous background composed of sea-clutter and external noise the application of Constant False Alarm Rate (CFAR) algorithms with a single parameter set are likel...
Conference Paper
The radar group at Helmut Schmidt University / University of the Federal Armed Forces Hamburg (HSU) is working in the field of coastal and mobile HF surface wave radar (SWR) since 2009. Supported by the Bundeswehr Technical Centre for Ships and Naval Weapons (WTD 71) in Eckernforde new signal processing techniques for clutter suppression and enhanc...
Conference Paper
The Boss SD-1 Super Overdrive effect pedal is a classical overdrive circuit for electric guitars. It consists of commonly found building blocks, namely common collector circuits for input and output buffering, the distortion stage as an op-amp circuit with diodes in the feed-back, and a tone-control block as a parametric linear circuit around an op...
Conference Paper
A new approach for the modeling of triodes is presented, featuring simple and physically-motivated equations. The mathematical description includes the replication of the grid current, which is a relevant parameter for the simulation of overdriven guitar amplifiers. If reference data from measurements of practical triodes is available, an individua...
Conference Paper
In order to distinguish between different signal paths, full orthogonal waveforms for each transmit antenna element is desirable for the concept of MIMO radar. We investigate time staggered Frequency Modulated Continuous Wave (FMCW) chirp signals with an individual successive temporal delay larger than the maximum round-trip time of the radar signa...
Conference Paper
This paper evaluates the performance of the time-staggered (TS) multiple-in-multiple-out (MIMO) Frequency Modulated Continuous Wave (FMCW) radar approach and compares it to a conventional phased-array approach to obtain a proof-of-concept. The approach combines stretch processed FMCW and colocated MIMO radar by using multiple time-staggered chirps....
Conference Paper
We investigate a low complexity Soft-Input Soft-Output (SISO) Hamming Decoder. The Decoding is based on error patterns which belong to the same syndrome. It is shown that it is sufficient to investigate error patterns with one and two errors to gain up to 1.35 dB compared to hard decision decoding. The proposed decoding algorithm has a linearly ris...
Conference Paper
Typical small-scale public address systems such as exhibition setups make use of video installation, audio playback and moderation to attract the attention of visitors. To reduce mutual interference between close positioned exhibition booths, organizers often restrain the sound pressure level. In this contribution we present a small public address...
Chapter
IntroductionDynamic range controlMusical distortion and saturation effectsExciters and enhancersConclusion Sound and musicReferences
Chapter
IntroductionModulatorsDemodulatorsApplicationsConclusion Sound and musicReferences
Chapter
Digital audio effects DAFX with MATLAB® Classifications of DAFX Fundamentals of digital signal processing Conclusion References
Conference Paper
Estimation of direction of arrival (DOA) of a radar backscatter signal by means of an amplitude comparison monopulse estimation scheme (ACMES) is a common high precision technique. But due to spatial undersampling in a sparse array architecture the conventional ACMES results in azimuth ambiguity. In this paper it is proposed that azimuth ambiguity...
Conference Paper
The development of a small, robust and cheap embedded hardware is presented, based on the TMS320C6722 signal processor. The system is tailored to the particular needs of audio and especially guitar effect processing, and is dedicated for educational use. The hardware realization, with focus on component selection, and the software framework are exp...
Conference Paper
Full-text available
Typical exhibition setups make use of video installation, audio playback and moderation to angle for the visitors attention. To reduce mutual interference between close positioned booths, organizers often restrain the sound pressure level. In this work we propose a small public address system meeting the requirements in the mentioned environments....
Conference Paper
Today flat panel loudspeakers are used in multiple applications. Due to their high directivity and their good structural integration properties, flat panel loudspeakers are commonly used for directed acoustic information. A previously proposed system of 2 parallel flat panel dipole loudspeakers with adapted input filtering ensures a high suppressio...
Article
Full-text available
Plane wave actuators without an enclosure per se have a forward and backward radiation. The backward radiation is unwanted in many applications when a single direction radiation is desired. To avoid the disadvantages of an enclosure a system is proposed, which provides a high suppression of the unwanted backward radiation using a pair of plane wave...
Conference Paper
Full-text available
The real-time simulation of analog circuits by digital systems becomes problematic when parametric components like potentiometers are involved. In this case the coefficients defining the digital system will change and have to be adapted. One common solution is to recalculate the coefficients in real-time, a possibly computationally expensive operat...
Conference Paper
A modified signal processing for FMCW waveform has been proposed that yields an improved range resolution without increasing the transmitted signal bandwidth. The chirp duration is reduced by an appropriate factor while the interval of the beat signal sequence considered for the range transform is kept constant. This modification implies a range tr...
Conference Paper
Most flat panel loudspeakers grant good structural integration properties due to the lack of an enclosure. However the backward radiation of a dipole is not desired in many applications. In this paper, a system which achieves a high suppression of the backward radiation using a pair of dipole speakers with adapted input signal filters, is proposed....
Conference Paper
Full-text available
Estimating room resonances in locations of big events and looking for counter-measures are normally done by sound engineers, mainly before the beginning of the event. In this paper an automation to enhance the audio quality in event rooms by suppressing the room resonances with a parametric equalizer of several high-Q peak filters is proposed. The...
Conference Paper
Full-text available
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.
Conference Paper
Full-text available
Measurement of impulse responses is a common task in audio signal processing. In this paper three common measurement techniques are reviewed: Maximum length sequences, exponentially swept sines and time delay spectrometry. The aim is to give the reader a brief tutorial of the methods with a special focus on deficiencies of the algorithms, aiding in...
Conference Paper
In audio codec design, often various parameters have to be fixed which may have a dramatic impact on codec performance. In this paper, we report on successful optimization of a codec based on perceptual criteria. Specifically, the PEAQ measure is used to determine the audio quality over a set of test items and search algorithms are used for optimiz...
Conference Paper
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-adapti...
Conference Paper
Considering high-resolution, multi-channel audio, it is worth re-examining the concept of psychoacoustically optimized noise shaping to ensure that signal quality is preserved when word-lengths are reduced. In this paper, approaches in static (time-invariant) and signal-adaptive (time-variant) noise shaping are discussed. We identify problems occur...
Conference Paper
We propose an extension of ADPCM that includes adaptive pre- and post-filtering to achieve spectral shaping of the coding noise. The advantage of this coding scheme is that it allows a realization without algorithmic delay by making the filters backwards-adaptive. The measurements we present indicate that the addition of adaptive pre- and post-filt...
Conference Paper
Full-text available
A straight-forward design of graphic equalizers with minimum-phase behavior based on recently developed higher-order band-shelving filters is presented. Due to the high filter order, the gain in one band is almost completely independent from the gain in the other bands. Although no special care will be taken to design filters with complementary edg...
Conference Paper
The main characteristic of shelving filters, as commonly used in audio equalization, is to amplify or attenuate a certain frequency band by a given gain. For parametric equalizers, a filter structure is desirable that allows independent adjustment of the width and center frequency of the band, and the gain. In this paper, we present a design for ar...
Conference Paper
Recursive shelving and peak filters are commonly used in audio equalization. Their main characteristic is to amplify or attenuate a certain frequency band by a given gain, while leaving signal components outside the respective band unaltered. For parametric equalizers, it is desirable to have a filter structure that allows for easy and independent...
Conference Paper
A new class of compander systems is proposed that combines conventional broad-band companders with adaptive filtering based on linear prediction. This allows not only for reduction, but also spectral shaping of noise induced e.g. in FM radio links. Evaluation using a simulation application shows a significant increase in perceived audio quality com...
Article
Although a lot of theoretical work has been done on purely functional data structures, few of them have actually been implemented to general usefulness, let alone as part of a data structure library providing a uniform framework.

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Projects (3)
Project
A Julia package for circuit simulation - https://github.com/HSU-ANT/ACME.jl