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Quantized Compressive Sampling for Structured Signal Estimation - Dr Niklas Koep - Bog - Verlag G. Mainz - Plusbog.dk

Quantized Compressive Sampling for Structured Signal Estimation - Dr Niklas Koep - Bog - Verlag G. Mainz - Plusbog.dk

This thesis investigates different approaches to enable the use of compressed sensing (CS)-based acquisition devices in resource-constrained environments relying on cheap, energy-efficient sensors. We consider the acquisition of structured low-complexity signals from excessively quantized 1-bit observations, as well as partial compressive measurements collected by one or multiple sensors. In both scenarios, the central goal is to alleviate the complexity of sensing devices in order to enable signal acquisition by simple, inexpensive sensors. In the first part of the thesis, we address the reconstruction of signals with a sparse Fourier transform from 1-bit time domain measurements. We propose a modification of the binary iterative hard thresholding algorithm, which accounts for the conjugate symmetric structure of the underlying signal space. In this context, a modification of the hard thresholding operator is developed, whose use extends to various other (quantized) CS recovery algorithms. In addition to undersampled measurements, we also consider oversampled signal representations, in which case the measurement operator is deterministic rather than constructed randomly. Numerical experiments verify the correct behavior of the proposed methods. The remainder of the thesis focuses on the reconstruction of group-sparse signals, a signal class in which nonzero components are assumed to appear in nonoverlapping coefficient groups. We first focus on 1-bit quantized Gaussian observations and derive theoretical guarantees for several reconstruction schemes to recover target vectors with a desired level of accuracy. We also address recovery based on dithered quantized observations to resolve the scale ambiguity inherent in the 1-bit CS model to allow for the recovery of both direction and magnitude of group-sparse vectors. In the last part, the acquisition of group-sparse vectors by a collection of independent sensors, which each observe a different portion of a target vector, is considered. Generalizing earlier results for the canonical sparsity model, a bound on the number of measurements required to allow for stable and robust signal recovery is established. The proof relies on a powerful concentration bound on the suprema of chaos processes. In order to establish our main result, we develop an extension of Maurey’s empirical method to bound the covering number of sets which can be represented as convex combinations of elements in compact convex sets.

DKK 445.00
1

Speech Signal Enhancement by Information Combining - Dr Florian Kurt Wolfgang Paul Heese - Bog - Verlag G. Mainz - Plusbog.dk

Speech Signal Enhancement by Information Combining - Dr Florian Kurt Wolfgang Paul Heese - Bog - Verlag G. Mainz - Plusbog.dk

Mobile phones as well as tablets are omnipresent and belong to everyday life. Today audiovisual communication takes place at different locations and in a large variety of acoustic environments. In consequence, the intelligibility as well as the quality of speech may significantly be degraded by ambient background noise. In order to improve speech intelligibility and to ensure a convenient communication with high audio quality, speech enhancement techniques are required. In this thesis all critical components contributing to the enhancement of the up-link signal are addressed: • signal capturing at the acoustic front-end with a new near field beamformer • new codebook based speech and noise estimation procedure generating and exploiting reliability information, and • actual noise reduction exploiting spectral dependencies of human speech. For the acoustic front-end of the digital processing chain a novel concept for the filter optimization of a near field beamformer is introduced. The optimization scheme allows to closely approximate a predefined reception characteristic which can be freely chosen according to the application. The output of the beamformer provides a pre-enhanced signal with improved SNR for subsequent single-microphone based speech enhancement. Single-microphone noise reduction usually relies on statistical properties of speech and noise. In general, the noise is assumed to be stationary or only slightly time-varying, which is in practice often not fulfilled. Due to imprecise noise estimation, single-microphone systems are prone to unpleasant artifacts that are called musical tones. In this context different Information Combining methods, merging various estimates, are presented which address specifically the problem of non-stationary noise signals, leading to a significant improved estimation accuracy. On the one hand, the proposed Information Combining is used with respect to spectral dependencies of human speech. On the other hand, it merges the best of several speech and noise estimates depending on their reliability. The necessary estimates are provided by a new statistical noise estimator as well as a codebook driven speech and noise estimation algorithm. The achieved estimation quality opens up the possibility to close the gap between the conflicting goals of high noise attenuation, low speech distortion, and the prevention of undesired musical tone artifacts. Finally, the practical aspects of the proposed enhancement systems are considered and discussed with two implemented real-time demonstrators.

DKK 445.00
1

Wind Noise Reduction - Dr Christoph Matthias Nelke - Bog - Verlag G. Mainz - Plusbog.dk

Wind Noise Reduction - Dr Christoph Matthias Nelke - Bog - Verlag G. Mainz - Plusbog.dk

With the technological progress, devices, such as mobile phones, tablet computers or hearing aids, can be used in a large variety of every-day situations for mobile communication. Acoustic background noise signals, which are picked up with the desired speech signal, can impair the signal quality and the intelligibility of a conversation. A special noise type is generated outdoors, if the microphone is exposed to a wind stream resulting in strong-rumbling noise, which is highly non-stationary. As a result, conventional approaches for noise reduction fail in the case of noise induced by wind turbulences. This thesis is focused on the development of signal processing concepts, which reduce the undesired effects of wind noise. The key contributions are: • Signal analysis of wind noise • Digital signal model for wind noise generation • Signal processing algorithms for detection and reduction of wind noise signals. All these topics are considered with the focus on the development of algorithms for single and dual microphone systems. The analysis of recorded wind signals is the first step and gives valuable information for the estimation and reduction of wind noise. Furthermore it leads to a signal model for the generation of reproducible artificial wind noise signals. For the enhancement of the disturbed speech, an estimate of the underlying wind noise signal is required. In contrast to state-of-the-art noise estimation algorithms, the spectral shape and energy distribution is exploited for the distinction between speech and wind noise components leading to a novel estimation scheme of the wind noise short-term power spectrum. Considering a system with two microphone inputs, the complex coherence function of the two recorded signals is exploited for wind noise estimation. In addition to commonly used noise reduction schemes by spectral weighting, an innovative concept for speech enhancement is developed by using techniques known from artificial bandwidth extension. Highly disturbed speech parts are replaced by corresponding parts from an artificial speech signal. Objective measures indicate a significant increase of both the signal-to-noise ratio and the speech intelligibility. Besides, two application examples show that the proposed methods are very efficient and robust in realistic scenarios.

DKK 445.00
1

Continuous-Amplitude Error Protection of Discrete-Time Signals - Dr Tim Schmitz - Bog - Verlag G. Mainz - Plusbog.dk

Continuous-Amplitude Error Protection of Discrete-Time Signals - Dr Tim Schmitz - Bog - Verlag G. Mainz - Plusbog.dk

Transmitting analog audio, video, or sensor data with a digital system requires sampling and quantization. While a sampled bandlimited signal can be reconstructed perfectly if the Nyquist-Shannon sampling theorem is met, quantization unavoidably adds irreversible errors. Given a certain bitrate for quantization (or source coding in general), the achievable signal quality is therefore limited. Transmission of the signal over a noisy channel causes additional errors. If some part of the bitrate is used for protecting the signal with a channel code, these additional errors can be reduced or even eliminated. However, this reduces the achievable maximum quality if the gross bitrate is fixed. Since this compromise between quantization and channel coding is often designed for the worst-case channel, the signal quality saturates early with increasing channel quality, and transmission over a better channel is far below optimum. Instead of conventional digital systems, this thesis covers a continuous-amplitude, discrete-time system. Such a system does not exhibit this saturation of the signal quality, as its performance further improves with increasing channel quality. An example of such a system is a channel coding strategy using analog modulo block codes (AMB codes) for error protection. The encoder multiplies the unquantized symbol vector with a code matrix and limits the transmission power with a modulo operation. In practice, AMB coding is implemented by digital signal processing. The digital representation of the information symbols can be very precise and no coarse quantization is required for bitrate reduction. In this thesis, AMB codes are analyzed, designed, and evaluated. The resulting code words, consisting of a lattice and a continuous component, are presented and analyzed. New decoding techniques are developed and evaluated. Additionally, a novel method to analytically estimate the quality of the decoded signal in terms of signal-to-distortion ratio (SDR) can replace computationally much more demanding simulations in many cases. This estimation strategy allows evaluating the usefulness of codes with different dimensionalities. A new, efficient design concept is developed for codes with a single input symbol, which are most promising in terms of complexity and performance. Asystem using AMBcodes outperforms traditional digital transmission systems with short block lengths for medium to high signal-to-noise ratios on the channel. Due to their typically very short block length and low encoding complexity, AMB codes are especially well suited for low-delay, low-power applications, such as hearing aids, wireless microphones, and wireless sensor networks.

DKK 445.00
1

Digital Enhancement of Speech Perception in Noisy Environments - Dr Markus Andreas Niermann - Bog - Verlag G. Mainz - Plusbog.dk

Digital Enhancement of Speech Perception in Noisy Environments - Dr Markus Andreas Niermann - Bog - Verlag G. Mainz - Plusbog.dk

In many everyday situations, people try to listen to speech from a loudspeaker, but the speech perception is affected by background noise. The location of a listener is called the near-end. Some applications in which this problem arises are mobile phones, public address systems, car entertainment systems, hearing aids, or head phones. Depending on the application, the speech may either be a recording, or, in case of communication systems, it may come live from a person speaking at the so-called far-end. Although it is usually not possible to cancel the near-end noise source, the speech perception can be enhanced nevertheless by adaptively pre-processing the loudspeaker signal. To find out in which way the speech should be pre-processed, it is beneficial to know the near-end noise characteristics and take them into account. The knowledge can be gained from a near-end microphone that is located close to the listener. This pre-processing technique is called Near-End Listening Enhancement (NELE). In this work several NELE concepts are developed and studied with the goal of enhancing the speech perception. Afterwards, a computationally very efficient and real-time capable NELE algorithm is developed, which can be implemented on signal processors and integrated into low-power devices. When deploying these enhancement concepts, additional problems may arise depending on the application. Two of them are studied in detail: 1. If the near-end loudspeaker and microphone are placed close to each other, which might occur at public address systems for instance, the speech from the loudspeaker is coupled into the near-end microphone as crosstalk and disturbs the estimation of the noise characteristics. An analysis of this arrangement shows that the functionality of the NELE algorithm is strongly affected. As a solution, it is proposed to use a new noise estimation technique which provides significantly more accurate noise estimates than state-of-the-art approaches by taking the adaptively predicted crosstalk into account. 2. Especially in the case of mobile communication, it is possible that a noise source is present also at the far-end. Consequently, the speech signal from the far-end is disturbed by noise. Typical mobile phones attenuate the far-end noise by employing techniques of noise reduction (NR) before transmitting the signal to the near-end. At the receiving near-end device, the denoised speech signal would then be processed by NELE to improve the speech perception, based only on the near-end noise. It is shown that NR and NELE counteract each other which results in unpleasant speech with strong artifacts. General solutions are proposed to perform NR and NELE under joint control or even jointly at the near-end or the far-end or within the network.

DKK 460.00
1

ODNP enhanced NMR relaxometry and diffusometry - Dr Till Uberruck - Bog - Verlag G. Mainz - Plusbog.dk

ODNP enhanced NMR relaxometry and diffusometry - Dr Till Uberruck - Bog - Verlag G. Mainz - Plusbog.dk

In nuclear magnetic resonance (NMR) the demand for compact, low-cost instruments that can substitute expensive superconducting magnets is growing. Although compact NMR devices based on permanent magnets that can resolve 1H chemical shift differences are commercially available, the magnetic field strength of these devices is limited, which sets boundaries to the signal intensity and quality. Hyperpolarization techniques to boost the NMR signal beyond ist thermally given polarization are well known and applied on superconducting magnets. Thus, implementing these methods on compact NMR instruments will be the next step of development to gain an increased signal quality and widen the range of applications for those magnets. One part of this thesis deals with the development of compact magnets for an X-Band Overhauser Dynamic Nuclear Polarization (ODNP) setup and the construction of the hyperpolarization hardware itself. Two concepts of small open-access magnets are presented, simulated, and experimentally tested that allow a direct shimming of the magnetic field by displacing magnet blocks without the need of further pole shoes or shim pieces. The superior design is transferred to a bigger magnet, which can fit EPR resonators and is the center piece of the ODNP hardware. For this hardware an ODNP amplification board is constructed and a commercially purchased EPR resonator is modified to include NMR coils. All components are adjusted to each other, the communication between them is established and the basic functionality of the hardware demonstrated. Additionally to the hardware construction, a fast field mapping method is introduced to characterize the detection volume of compact NMR devices. This method facilitates the characterization and hence construction of permanent NMR magnets and saves about one order of magnitude in measurement time compared to the established procedures. Beside the construction of compact hyperpolarization setups, new applications areas for these techniques should be explored. In this context the idea of ODNP enhanced Laplace NMR experiments is explained and experimentally demonstrated by CPMG, inversion recovery, and PFG diffusion experiments on a model sample. Furthermore, these techniques are applied to study the influence of ODNP spin probes on the dynamic properties of Nafion membranes. Since ODNP relies on the presence of paramagnetic spin probes their influence on the material properties must be studied before any conclusion on material properties can be drawn from hyperpolarization experiments. A cornucopia of established and novel methods is applied to dissect Nafion properties in the presence of spin probes ranging from Small Angle X-ray Scattering (SAXS), Thermal Gravimetric Analysis (TGA), conductivity measurements, PFG NMR diffusometry, field cycling NMR relaxometry, ODNP relaxometry to Electron Paramagnetic Resonance (EPR), and new combinations thereof.

DKK 460.00
1

Active Noise and Occlusion Effect Cancellation in Headphones and Hearing Aids - Dr Stefan Wilhelm Liebich - Bog - Verlag G. Mainz - Plusbog.dk

Active Noise and Occlusion Effect Cancellation in Headphones and Hearing Aids - Dr Stefan Wilhelm Liebich - Bog - Verlag G. Mainz - Plusbog.dk

The perception of one’s own voice is distorted when telephoning with headsets, or wearing hearing aids. The reason for this is the so-called occlusion effect, which occurs when ear canals are completely or partially closed by the headset or hearing aid. The occlusion causes amplification at low frequencies, and attenuation at high frequencies of one’s own voice. The unnatural perception of one’s own voice and of noise caused by chewing and swallowing are among the most common complaints of users. Furthermore, environmental noise might impair perception. In this thesis, both the unnatural perception of one’s own voice and the disturbance by environmental noise are tackled by a novel signal processing approach. The proposed solution solves the problem of the occlusion effect by actively emitting a compensation signal through the integrated loudspeaker. The approach is called Occlusion Effect Cancellation (OEC) and significantly improves the perception of one’s own voice and of the acoustic environment. This novel approach combines methods of active noise cancellation (ANC, Noise Cancelling Headphone) with a personalized design. The bilateral headset contains two additional microphones per side, one inner and one outer, to acquire signals for the calculation of the compensation signals. A correctly balanced processing of the two microphone signals results in a "digital ear opening" and a much more natural perception of both one’s own voice and of the environment. The extent of the digital ear opening is controllable. The system can also be operated as a noise cancelling headphone by changing the parameters to a conventional design to create an acoustic isolation from the environment. This thesis proposes a novel robust approach based on digital filtering to solve the described problems. A combination of feedback and feedforward filter design allows for either approaching personal silence or a natural perception of one’s own voice and the acoustic environment.

DKK 462.00
1

HD Telephony by Artificial Bandwidth Extension - Dr Thomas Matthias Schlien - Bog - Verlag G. Mainz - Plusbog.dk

HD Telephony by Artificial Bandwidth Extension - Dr Thomas Matthias Schlien - Bog - Verlag G. Mainz - Plusbog.dk

The audio bandwidth of digital landline and mobile telephone networks is still mostly restricted to 200 Hz to 3.4 kHz. This is due to compatibility requirements during the transition phase from analogue to digital transmission technology. The resulting characteristic "telephone speech" is widely accepted, but the intelligibility of syllables is only 91%. Meanwhile, improved coding standards for so-called “HD voice” or “Wideband Speech” have been developed which are gradually being introduced into the networks. They support an audio frequency bandwidth of 50 Hz to 7.0 kHz with significantly increased audio quality and speech intelligibility. For a very long time however, new HD-telephones and old narrowband telephones have to co-exist. If an HD-terminal is connected over a narrowband link to an old telephone, the improved coding scheme cannot be used. In this thesis, signal processing concepts are developed for improving audio quality and intelligibility of narrowband speech by artificial bandwidth extension (ABWE). These algorithms can be applied in the HD terminals or in the network, to transform narrowband speech to HD voice. Based on the source-filter model of speech production and a priori knowledge of the characteristics of speech signals, the missing frequency components between 3.4 kHz and 7 kHz are reconstructed. In comparison to the state-of-the-art ABWE approaches, the main contributions are: • new concepts of estimating the wideband spectral envelope, e.g., in terms of the model filter by interpolation in the acoustic tube domain • algorithms for spectral extension of the excitation signal • new insights concerning the relative importance of the excitation, the temporal envelope and the spectral envelope • remarkable improvements of the audio quality • significant low computational complexity • efficient and effective training and estimation algorithms The improvements are verified by objective evaluation and subjective listening tests.

DKK 445.00
1

Studies of Tangible Cultural Heritage with Portable Stray-Field NMR - Dr Christian Rehorn - Bog - Verlag G. Mainz - Plusbog.dk

Studies of Tangible Cultural Heritage with Portable Stray-Field NMR - Dr Christian Rehorn - Bog - Verlag G. Mainz - Plusbog.dk

Sensors for stray-field nuclear magnetic resonance (NMR) have been employed to measure tangible cultural heritage since they were first conceived. Although NMR is a method with inherently low sensitivity and requires larger amounts of time than many other techniques, it is non-destructive and grants access to spin densities and relaxation times, physical quantities which are exclusive to the method. This thesis describes theory, instrumentation, and applications of unilateral NMR in the field of cultural heritage. The detection zone of an NMR-MOUSE, a sensor developed and maintained by past and present members of the group of Professor Blümich, is mapped through back-projection for the first time. Traditionally, the signal arising from the detection volume was averaged over multiple scans until the signal response was satisfactory. In this work, an algorithm is proposed to improve the signal-to-noise ratio in magnetization decays as a method of post processing an output, which contains the individual data from smaller sub-experiments. One of the most prominent fields of application for unilateral NMR is the investigation of porosity in stone and soil. In a study of ancient mortars in Herculaneum, a city buried by the ashes of Mount Vesuvius during the famous eruption of 79 AD, the profiles of over 60 sites, fragments and mock-ups were compared for the first time with methods known from statistics and pattern recognition. The effects of high temperatures up to 900 °C on low and high-density sandstone were determined in terms of their transverse relaxation times. Mock-ups were cross referenced with actual walls and fragments to assess the damage of the western wall of the burnt down Mackintosh Library in Glasgow. Studies of paint show how stray-field sensors can help evaluating solvent activity and their potential as cleaning agents. Relaxation times assess embrittlement and transient softening caused by such treatments. Furthermore, the potential to detect or even discriminate natural and artificial aging of binder polymers is explored. The impressive sound and quality of historic instruments by Stradivari and his contemporaries is still unsurpassed in the opinion of many musicians, but the reason for this remains controversial. When tested with unilateral NMR, violins and violas from the golden age of luthiers revealed a homogeneous wood density throughout their maple back plate. Shorter relaxation times at the wood surfaces may be traces of previous treatments. Even though unilateral NMR equipment was claimed to be portable for many years, it had relied on grid power and shielding from weather. This work presents the efforts of measuring biofilms in Yellowstone National Park, relying on battery power to perform the non-destructive tests. An NMR-MOUSE was constructed and waterproofed to detect biofilms under water and in-situ.

DKK 445.00
1

Advances in NMR spectoscopy and imaging - Dr Alexander Gorges - Bog - Verlag G. Mainz - Plusbog.dk

Advances in NMR spectoscopy and imaging - Dr Alexander Gorges - Bog - Verlag G. Mainz - Plusbog.dk

Nuclear Magnetic Resonance (NMR) techniques have proliferated in many fields of science and technology like bio-sensing, chemical reaction monitoring and material characterization. Since the inception of NMR as an analytical tool, improving the sensitivity by increasing the field strength has been the primary development goal. However, in order to reduce cost and environmental impact, the trend to miniaturized NMR devices and its diverse application fields enjoys increasing interest. The first part of this thesis introduces novel insights into low-power rf-excitation, which is one crucial aspect for enabling further development in this direction, by employing Frank sequences. Based on experimental data, a detailed evidence of the power savings by excitation in the linear regime is given aiming at future elimination of the rf-amplifier from the NMR spectrometer so as to allow further mobility improvements. Selective excitation by colored Frank-sequence is reported, which bears promise for solvent signal suppression and motion tagging in magnetic resonance imaging (MRI). To this end, spectroscopic quality as well as image resolution with Frank excitation was significantly improved. The aim of the second part is to provide quantitative 3D moisture content patterns of natural soil samples on the microscale, which are essential for improving geological simulations on the field scale. In the course of this, the standardized imaging sequences ’zero echo time’ (ZTE) and ’ultra-short echo time’ (UTE) were employed but also Frank-sequence excitation was implemented, reflecting its first genuine application. In order to characterize water-retention behavior of the soil samples, ZTE experiments were combined with standardized geological methods like Multi-Step Outflow (MSO) and tensiometric measurements. Compared to the established way of acquiring water-retention data, the introduced method provides a fast and precise method with low effort.

DKK 445.00
1

Physically-Based Models for the Analysis of Raman Spectra - Dr Peter Beumers - Bog - Verlag G. Mainz - Plusbog.dk

Physically-Based Models for the Analysis of Raman Spectra - Dr Peter Beumers - Bog - Verlag G. Mainz - Plusbog.dk

In recent years, spectroscopy has developed into an increasingly valuable tool to determine the composition of mixtures; for scientific questions as well as for the industry. The increasing use of spectroscopy raises the question how to best use the obtained data. For the analysis of spectral data, the method of Indirect Hard Modeling (IHM) has been established besides statistical methods like PLS. IHM is a nonlinear method that can therefore efficiently treat nonlinear effects such as peak-shifts. In the present work, the IHM method is expanded to increase its applicability. IHM treats nonlinear effects in the spectral evaluation. Therefore, the direct proportionality between the concentration and the Raman signal of a component can be used for calibration. The resulting linear calibration model allows for reliable extrapolation. Thus, IHM can be used to study reactive systems, even if only binary subsystems can be used for calibration. However, thermodynamic systems with intermediates can so far only be calibrated by using thermodynamic models. In this work, a method is established that calibrates a reactive system with intermediates only based on the reaction mechanism as well as stoichiometry and electroneutrality. Spectral backgrounds, e.g., fluorescence, can be treated by a spectral pretreatment or via background models. However, spectral backgrounds are still a common source of error in IHM. Derivatives have long been used very effectively in statistical methods. Therefore, IHM is adapted so that it becomes possible to evaluate the first derivative of spectra. The calibration of IHM is mostly limited to the relative spectral intensities of the involved components. In the present work, a method is presented that uses the information in the calibration spectra more thoroughly. For this purpose, nonlinear effects are parametrized as a function of concentration. The commonly used peak profiles do not reflect the physical reality at a detector very well. As a result, narrow modelled peaks may change their apparent intensity if they are shifted. To correct these shortcomings, a new peak model is proposed in this work that is more closely aligned to the physical reality of a detector.

DKK 460.00
1

A Milliliter-Scale Setup for the Efficient Characterization of Multicomponent Vapor-Liquid Equilibria Using Raman Spectroscopy - Dr Bastian

A Milliliter-Scale Setup for the Efficient Characterization of Multicomponent Vapor-Liquid Equilibria Using Raman Spectroscopy - Dr Bastian

Vapor-liquid equilibrium (VLE) data are of major importance for the chemical industry. Despite significant progress in predictive methods, experimental VLE data are still indispensable. In this work, we address the need for experimental VLE data. Commonly, the characterization of VLE requires significant experimental effort. To limit the experimental effort, VLE measurements are frequently conducted by synthetic methods which employ samples of known composition and avoid complex analytics and sampling issues. In contrast, analytical methods provide independent information on phase compositions, commonly based on sampling and large amounts of substance. In the first part of this work, we employ a synthetic method, the well-established Cailletet setup, to characterize the high pressure VLE of two promising binary biofuel blends. The Cailletet method serves as a state of the art reference method that enables collecting data of remarkable accuracy. However, extensive infrastructure is needed. In the second part, to avoid extensive infrastructure and overcome limitations of previous methods, we develop a novel analytical milliliter-scale setup for the noninvasive and efficient characterization of VLE: RAMSPEQU (Raman Spectroscopic Phase Equilibrium Characterization). The novel setup saves substance and rapidly characterizes VLE. Sampling and its associated errors are avoided by analyzing phase compositions using Raman spectroscopy. Thereby, volumes of less than 3 ml are sufficient for reliable phase equilibrium measurements. To enable rapid data generation and save substance, we design an integrated workow combining Raman signal calibration and VLE measurement. As a result, RAMSPEQU gives access to up to 15 pT xy-data sets per workday. RAMSPEQU is successfully validated against pure component and binary VLE data from literature. However, mixtures with only two components rarely depict real industrial applications. As the number of experiments increases strongly with a rising number of components, the efficient RAMSPEQU setup seems particularly suited for multicomponent systems. In the third part of this work, we employ the RAMSPEQU setup for the characterization of a quaternary system and its binary subsystems. 22 ml and 105 ml of the binary and quaternary mixtures are sufficient for an extensive VLE characterization. The RAMSPEQU setup and its integrated workow enable the characterization of multicomponent VLE while saving significant amounts of substance and laboratory time.

DKK 460.00
1