This paper addresses the problem of automatic monitoring the weld quality produced by robotised short arc welding. A simple statistical change detection algorithm for the weld quality, recursive Sequential Probability Ratio Test (SPRT), is used. The algorithm may equivalently be viewed as a cumulative sum (CUSUM) - type test. The test statistics is based upon the variance of the amplitude of the weld voltage. The performance of the algorithm is evaluated using experimental data. The results obtained from the algorithm indicate that it is possible to detect changes in the weld quality automatically and on-line.

This paper deals with the problem of automatic monitoring the weld quality when welding with Gas Metal Arc (GMA) in short circuiting mode. Experiments with two different types of T-joints are performed in order to provoke optimal and non-optimal welding conditions. During the experiments, voltage and current are measured from the welding process. A simple statistical change detection algorithm for the weld quality, the repeated Sequential Probability Ratio Test (SPRT), is used. The algorithm can equivalently be viewed as a cumulative sum (CUSUM) - type test. The test statistics is based upon the fluctuations of amplitude in the weld voltage. It is shown that the fluctuations of the weld voltage amplitude decreases when the welding process is not operating under optimal condition. The results obtained from the experiments indicate that it is possible to detect changes in the weld quality automatically and on-line.

This paper addresses the problem of automatic monitoring the weld quality produced by robotised short arc welding. A simple statistical change detection algorithm for the weld quality, recursive sequential probability ratio test (SPRT), is used. The algorithm may equivalently be viewed as a cumulative sum (CUSUM) type test. The test statistics is based upon the variance of the amplitude of the weld voltage. It is shown that the variance of the weld voltage amplitude decreases when the welding process is not operating under optimal condition. The performance of the algorithm is evaluated using experimental data. The results obtained from the algorithm indicate that it is possible to detect changes in the weld quality automatically and on-line

This paper addresses the problems involved in the automatic monitoring of the weld quality produced by robotized short-arc welding. A simple statistical change detection algorithm for the weld quality, the repeated Sequential Probability Ratio Test (SPRT), was used. The algorithm may similarly be viewed as a cumulative sum (CUSUM) type test, and is well-suited to detecting sudden minor changes in the monitored test statistic. The test statistic is based on the variance of the weld voltage, wherein it will be shown that the variance decreases when the welding process is not operating under optimal conditions. The performance of the algorithm is assessed through the use of experimental data. The results obtained from the algorithm show that it is possible to detect changes in weld quality automatically and on-line.

This paper addresses the problem of automatic detection of burn-through in weld joints. Gas metal are (GMA) welding with pulsed current is used, and welding voltage and current are recorded. As short-circuitings are common between the welding electrode and the work piece during burn-through, a short-circuit detector is developed to detect these events. To detect another specific characteristic of burn-through-a broadband long-lasting voltage component-this detector is combined with a square-law detector. This second detector is based on a non-linear modification of an autoregressive model with extra input (ARX-model) of the welding process. The results obtained from this compound detector indicate that it is possible to detect burn-through in the welds automatically. The work also indicates that it is possible to design an on-line monitoring system for robotic GMA welding.

Denna rapport analyserar ekosläckning för en full duplex högtalartelefon. Systemet bygger på en LMS-algoritm som har implementerats på en TMS320C3 1 signalprocessor. Arbetet har gått ut på att undertrycka ekon genom utnyttja så mycket som möjligt av processoms kapacitet. Möjligheterna för att två eller fler personer skall kunna tala med varandra på att ledigt sätt genom en högtalartelefon påverkas mycket av de ekon som finns i systemet. Högtalartelefonens mikrofon fångar upp ljudet ifrån sin egen högtalare och skickar detta ljud tillbaka genom telefonnätet. Utan någon form av ekokontroll uppstår ett irriterande eko av sin egen röst i telefonen. Ju längre fardröjning som telefonnätet tillför, ju mer irriterande blir ekot. Konventionella högtalartelefoner använder röstaktivering av mikrofonen för att bli av med ekot. Detta medför att de samtalande partertema inte både kan höra och prata med varandra samtidigt. Detta kallas i kommunikationssammanhang Halv-Duplex. Denna onaturliga form av samtal medför att bägge börjar tala samtidigt och att de avbryter varandra. 1 mitt arbete tittar jag närmare på metoder för ekokontroll med mål att bägge parterna skall både kunna tala med och höra varandra simultant, sk. Full-Duplex, utan rundgång. Teorierna kring adaptiv ekosläckning har varit kända sedan Iänge men det är inte förrän på senare tid som det finns processorer som klarar av beräkningarna’ i realtid. TMS320C3 1 ‘an har inte riktigt den önskvärda beräkningskapaciteten för realtidsberäkning av langa adaptiva FIR-filter. En filterlängd på 300 filter koefficienter uppnåddes. För att nå upp till denna längd användes en metod med adaptering bara på halva bandbredden, 1500 Hz. Den andra halvan av bandbredden, 1500 - 3500 Hz, i ett telefonnät lämnades ofiltrerad. Ett stort bekymmer visade sig vara att finna en lamplig metod för att besluta när adaptering får ske. En enkel sk. taldetektor implementerades för detta endamål.

In the turning operation the relative dynamic motion between cutting tool and workpiece, or vibration is a frequent problem, which affects the result of the machining, in particular the surface finish. The tool life is also influenced by the vibrations. When the working environment is considered, noise is frequently introduced by dynamic motion between the cutting tool and the workpiece. By proper machine design, e.g. improved stiffness of the machine structure, the problem of relative dynamic motion between cutting tool and workpiece may be partially solved. However, by active control of machine-tool vibration, a further reduction of the dynamic motion between cutting tool and workpiece can be achieved. It was found that adaptive feedback control based on the filtered-x LMS-algorithm enables a reduction of the vibration with up to 40 dB at 1.5 kHz and simultaneously with approximately 40 dB at 3 kHz. A significant improvement of the workpiece surface was observed and a substantial improvement of the acoustic noise level was obtained with adaptive control.

In the turning operation the relative dynamic motion between cutting tool and workpiece, or vibration, is a frequent problem, which affects the result of the machining, and in particular, the surface finish. Tool life is also influenced by vibration. Noise in the working environment frequently occurs as a result of dynamic motion between the cutting tool and the workpiece. With proper machine design, i.e. improved stiffness of the machine structure, the problem of relative dynamic motion between cutting tool and workpiece may be partially solved. However, by active control of machine-tool vibration, a further reduction of the dynamic motion between cutting tool and workpiece can be achieved. It was found that adaptive feedback control of tool vibration in the cutting speed direction, based on the filtered-x LMS-algorithm, enables a reduction in vibration, by up to 40 dB at 1.5 kHz, and by approximately 40 dB at 3 kHz. It was also observed that the introduction of leakage in the filtered-x LMS-algorithm improved the stability properties of the feedback control system. A significant improvement in the workpiece surface was observed and a substantial improvement in the acoustic noise level was obtained with adaptive control.

In the turning operation the relative dynamic motion between cutting tool and workpiece, or vibration, is a frequent problem, which affects the result of the machining, and, in particular, the surface finish. Tool life is also influenced by vibration. Severe acoustic noise in the working environment frequently occurs as a result of dynamic motion between the cutting tool and the workpiece. These problems can be reduced substantially by active control of the machine-tool vibration. Adaptive feedback control based on the filtered-x LMS-algorithm, enables a reduction of the vibration by up to 40 dB at 1.5 kHz and by approximately 40 dB at 3 kHz. The active control performeds a broadband attenuation of the sound pressure level by up to 35 dB. However, the process of machining a workpiece usually involves a variety of cutting data which in turn are likely to cause substantial variations in the spectral properties of the tool vibrations. Hence, variations in the spectral properties originates from changes in the excitation of the tool holder and changes in the structural response of the tool holder. To handle the potential large variations in the spectral properties of tool vibration in the turning operation the robustness of the control system has to be improved. By applying the leaky version of the filtered-x LMS algorithm in the active control of machine tool vibration it was found that the robustness of the adaptive control system was improved substantially to large variations in the spectral properties of tool vibration.

This communication treats the problem of controlling the superresolution in adaptive beamformers. A straightforward method is presented that works for both narrow-band and broad-band arrays. The method is based on forming the blocking matrix in a general sidelobe canceller (GSC) structure using a spatial FIR filter. The suppression of this spatial filter and the implicit noise of the leaky (LMS) algorithm together determine the beamformer.

Hands free input of mobile telephones is often almost impossible due to the noise situation in the car and the channel quality. The authors present a method to enhance the quality using an adaptive microphone array. Measurements show an improvement of the input with approximately 10 dB. Tape results are available for subjective evaluation

Hands free input of mobile telephones is often almost impossible due to the noise situation in the car and the channel quality. The authors present a method to enhance the quality using an adaptive microphone array. Measurements show an improvement of the input with approximately 10 dB. Tape results are available for subjective evaluation.

An implementation of a broadband adaptive array on a multiprocessor digital signal processing (DSP) system for use in a hands-free mobile radio telephone is described. This implementation of a five-microphone adaptive Griffiths-Jim array can handle FIR filters with up to 128 taps behind each microphone at a sampling rate of 8 kHz. The filter structure makes it possible to combine an adaptive array with a noise canceler. The near-field problem has been solved by using focusing, a speech-controlled adaptive algorithm, and a short hourglass. Preliminary measurements indicate a considerable potential for this technique in hands-free mobile telephony. The array gives a 20-30-dB suppression of a broadband jammer covering 300-1100 Hz, even with three reflecting walls surrounding the microphone.

The problem of finding the Wiener filters for a wideband adaptive beamformer is treated. Explicit expressions are given for the filters and the error power spectrum in the frequency domain. The expressions are simple to program and make it possible to investigate superresolution for wideband signals and to determine sufficient lengths of finite-impulse-response filters in adaptive arrays.

This Licentiate thesis is divided into three parts corresponding to three different papers. There is one research report, one conference paper and one submitted journal paper. All three parts deal with acoustic echo and/or noise cancelling problems when using adaptive microphone arrays. In particular, the papers address the performance of an adaptive microphone array in a small enclosure such as the car cabin. A calibrating scheme is proposed which is independent of array geometry and channel matching, and which calibrates the adaptive array to the given acoustic environment and to the given electronic equipment. Results from real measurements in a car interior are included and compared with an analytical description of an adaptive microphone array. Part A gives an analytical description of an adaptive microphone array which facilitates a simple built-in calibration to the environment and instrumentation. Part B describes the method for performing acoustic echo cancelling with a dig-ital "on-site", "self-calibrating" microphone array system. The calibration process is a simple indirect calibration which continuously adapts to the actual environment and electronic equipment. There is a US patent based on this part and an international patent is currently under examination. Part C presents a neural network based microphone array system, which is capable to continuously perform speech enhancement and adaptation to nonuniform quantization, such as A-law and µ-law.

This Licentiate thesis is divided into three parts corresponding to three different papers. There is one research report, one conference paper and one submitted journal paper. All three parts deal with acoustic echo and/or noise cancelling problems when using adaptive microphone arrays. In particular, the papers address the performance of an adaptive microphone array in a small enclosure such as the car cabin. A calibrating scheme is proposed which is independent of array geometry and channel matching, and which calibrates the adaptive array to the given acoustic environment and to the given electronic equipment. Results from real measurements in a car interior are included and compared with an analytical description of an adaptive microphone array. Part A gives an analytical description of an adaptive microphone array which facilitates a simple built-in calibration to the environment and instrumentation. Part B describes the method for performing acoustic echo cancelling with a dig-ital "on-site", "self-calibrating" microphone array system. The calibration process is a simple indirect calibration which continuously adapts to the actual environment and electronic equipment. There is a US patent based on this part and an international patent is currently under examination. Part C presents a neural network based microphone array system, which is capable to continuously perform speech enhancement and adaptation to nonuniform quantization, such as A-law and µ-law.

This paper presents a neural network based microphone array system, which is capable to continuously perform speech enhancement and adaptation to nonuniform quantization, such as A-law and $mu@-law. Such a quantizer is designed to increase the Signal to Quantization Noise Ratio (SQNR) for small amplitudes in telecommunications systems. The proposed method primarily developed for hands-free mobile telephones, suppresses the ambient car noise with approximately 10 dB. The system is based upon a multi-layered nonlinear back-propagation trained network by using a built-in calibration technique.

This report describes a novel method to perform acoustic echo cancelling with microphone arrays. The method employs a digital self-calibrating microphone system. The on-site calibration process is a simple indirect calibration which adapts in each special case to the environment and electronic equipment. The method also continuously takes into account environmental disturbances such as car engine noise and fan noise. The method is primarily aimed at handsfree mobile telephones, by suppressing the handsfree loudspeaker and car noise simultaneously. The report also contains an extensive evaluation in a car.

A novel method of performing acoustic echo canceling using microphone arrays is presented. The method employs a digital self-calibrating microphone system. The calibration process is a simple indirect on-site calibration that adapts to the particulars of the acoustic environment and the electronic equipment in use. Primarily intended for handsfree telephones in automobiles, the method simultaneously suppresses the handsfree loudspeaker and car noise. The system also continuously takes into account disturbances such as fan noise. Examples from an extensive evaluation in a car are also included. Typical performance results demonstrate 20-dB echo cancellation and 10-dB noise reduction simultaneously.

This paper presents a new practical approach to complex Chebyshev approximation by semi-infinite linear programming. The approximation problem may be general with arbitrary complex basis functions. By the new front-end technique, the associated semi-infinite linear programming problem is solved exploiting the finiteness of the related Lagrange multipliers by adapting finite--dimensional linear programming to the dual semi--infinite problem, and thereby taking advantage of the numerical stability and efficiency of conventional linear programming software packages. Furthermore, the optimization procedure is simple to describe theoretically and straightforward to implement in computer coding. The new design technique is therefore highly accessible. The new algorithm is formally introduced as the linear Dual Nested Complex Approximation (DNCA) algorithm. The DNCA algorithm is versatile and can be applied to a variety of applications such as narrow-band as well as broad-band beamformers with any geometry, conventional Finite Impulse Response (FIR) filters, analog and digital Laguerre networks, and digital FIR equalizers. The proposed optimization technique is applied to several numerical examples dealing with the design of a narrow-band base-station antenna array for mobile communication. The flexibility and numerical efficiency of the proposed design technique are illustrated with these examples where hundreds of antenna elements are optimized without numerical difficulties.

This paper presents a method to simultaneously perform 20 dB acoustic echo cancellation and 15-20 dB speech enhancement using an adaptive microphone array combined with spectral subtraction. Primarily intended for handsfree telephones in automobiles, the microphone array system simultaneously emphasizes the near-end talker and suppresses the handsfree loudspeaker and the broadband car noise. The array system is based on a fast and efficient on-site calibration and can be used in other situations such as conventional speaker phones.

This paper presents a new method to simultaneously perform 20 dB acoustic echo cancellation and 10 dB speech enhancement utilizing an adaptive microphone array. The system is based on a fast and efficient on-site calibration. Primarily intended for handsfree telephones in automobiles, the microphone array system simultaneously emphasizes the near-end talker and suppresses the handsfree loudspeaker and car noise. The method can also be used in other situations such as conventional speaker phones.

We consider a class of FIR filters defined by the first order difference routing digital filter (DRDF) structure and sums of two powers-of-two coefficients. A novel design method is developed for constructing high quality filters with reference to the min-max error criterion. This method is highly efficient in terms of computational time. Simulation studies show a large improvement over existing methods such as quantization. In some cases, the peak ripple magnitude over the stop and pass bands is reduced by up to 13 dB over the quantization method. These results are achieved even for cases involving small number of delays.

Conventional approches to the problem of extracting a clock from the data do not automatically hold the clock in the center of the data-eye. This thesis describes a data feedback technique that adjusts the clock and significantly reduces timing uncertainty by compensating for initial ciruit misalignment, propagation delay variations and low-frequency jitter. The purpose of this thesis is to investigate the characteristics of the technique with different phase detectors by analysis, simultations and experimentally work.