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  • 151.
    Chen, Jiandan
    et al.
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap, Avdelningen för elektroteknik.
    Khatibi, Siamak
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap, Avdelningen för elektroteknik.
    Kulesza, Wlodek
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap, Avdelningen för elektroteknik.
    Depth reconstruction uncertainty analysis and improvement: The dithering approach2010Ingår i: Image and Vision Computing, ISSN 0262-8856, E-ISSN 1872-8138, Vol. 29, nr 9, s. 1377-1385Artikel i tidskrift (Refereegranskat)
    Abstract [en]

    The depth spatial quantization uncertainty is one of the factors which influence the depth reconstruction accuracy caused by a discrete sensor. This paper discusses the quantization uncertainty distribution, introduces a mathematical model of the uncertainty interval range, and analyzes the movements of the sensors in an Intelligent Vision Agent System. Such a system makes use of multiple sensors which control the deployment and autonomous servo of the system. This paper proposes a dithering algorithm which reduces the depth reconstruction uncertainty. The algorithm assures high accuracy from a few images taken by low-resolution sensors. The dither signal is estimated and then generated through an analysis of the iso-disparity planes. The signal allows for control of the camera movement. The proposed approach is validated and compared with a direct triangulation method. The simulation results are reported in terms of depth reconstruction error statistics. The physical experiment shows that the dithering method reduces the depth reconstruction error.

  • 152. Chen, Jiandan
    et al.
    Khatibi, Siamak
    Kulesza, Wlodek
    Planning of a Multi Stereo Visual Sensor System Depth Accuracy and Variable Baseline Approach2007Konferensbidrag (Refereegranskat)
  • 153. Chen, Jiandan
    et al.
    Khatibi, Siamak
    Kulesza, Wlodek
    Planning of a Multi Stereo Visual Sensor System for a Human Activities Space2007Konferensbidrag (Refereegranskat)
  • 154. Chen, Jiandan
    et al.
    Khatibi, Siamak
    Wirandi, Jenny
    Kulesza, Wlodek
    Planning of a Multi Sensor System for Human Activities Space – Aspects of Iso-disparrity Surface2007Konferensbidrag (Refereegranskat)
    Abstract [en]

    The Intelligent Vision Agent System, IVAS, is a system for automatic target detection, identification and information processing for use in human activities surveillance. This system consists of multiple sensors, and with control of their deployment and autonomous servo. Finding the optimal configuration for these sensors in order to capture the target objects and their environment to a required specification is a crucial problem. With a stereo pair of sensors, the 3D space can be discretized by an iso-disparity surface, and the depth reconstruction accuracy of the space is closely related to the iso-disparity curve positions. This paper presents a method to enable planning the position of these multiple stereo sensors in indoor environments. The proposed method is a mathematical geometry model, used to analyze the iso-disparity surface. We will show that the distribution of the iso-disparity surface and the depth reconstruction accuracy are controllable by the parameters of such model. This model can be used to dynamically adjust the positions, poses and baselines lengths of multiple stereo pairs of cameras in 3D space in order to get sufficient visibility and accuracy for surveillance tracking and 3D reconstruction. We implement the model and present uncertainty maps of depth reconstruction calculated while varying the baseline length, focal length, stereo convergence angle and sensor pixel length. The results of these experiments show how the depth reconstruction uncertainty depends on stereo pair’s baseline length, zooming and sensor physical properties.

  • 155. Chen, Jiandan
    et al.
    Mustafa, Wail
    Siddig, Abu Bakr
    Kulesza, Wlodek
    APPLYING DITHERING TO IMPROVE DEPTH MEASUREMENT USING A SENSOR-SHIFTED STEREO CAMERA2010Ingår i: Metrology and Measurement Systems, ISSN 0860-8229, Vol. 17, nr 3Artikel i tidskrift (Refereegranskat)
    Abstract [en]

    The sensor-shifted stereo camera provides the mechanism for obtaining 3D information in a wide field of view. This novel kind of stereo requires a simpler matching process in comparison to convergence stereo. In addition to this, the uncertainty of depth estimation of a target point in 3D space is defined by the spatial quantization caused by the digital images. The dithering approach is a way to reduce the depth reconstruction uncertainty through a controlled adjustment of the stereo parameters that shift the spatial quantization levels. In this paper, a mathematical model that relates the stereo setup parameters to the iso-disparities is developed and used for depth estimation. The enhancement of the depth measurement accuracy for this kind of stereo through applying the dithering method is verified by simulation and physical experiment. For the verification, the uncertainty of the depth measurement using dithering is compared with the uncertainty produced by the direct triangulation method. A 49% improvement of the uncertainly in the depth reconstruction is proved.

  • 156. Chen, Jiandan
    et al.
    Olayanju, Iyeyinka Damilola
    Ojelabi, Olabode Paul
    Kulesza, Wlodek
    RFID Multi-Target Tracking Using the Probability Hypothesis Density Algorithm for a Health Care Application2011Konferensbidrag (Refereegranskat)
    Abstract [en]

    The intelligent multi-sensor system is a system for target detection, identification and information processing for human activities surveillance and ambient assisted living. This paper describes RFID multi-target tracking using the Gaussian Mixture Probability Hypothesis Density, GM-PHD, algorithm. The multi target tracking ability of the proposed solution is demonstrated in a simulation and real environment. A performance comparison of the Levenberg-Marquardt algorithm with and without the GM-PHD filter shows that the GM-PHD algorithm improves the accuracy of tracking and target position estimation significantly. This improvement is demonstrated by a simulation and by a physical experiment.

  • 157.
    Chen, Rongrong
    et al.
    Blekinge Tekniska Högskola, Sektionen för teknik, Avdelningen för signalbehandling.
    Zhu, Min
    Blekinge Tekniska Högskola, Sektionen för teknik, Avdelningen för signalbehandling.
    Birth Density Modeling in Multi-target Tracking Using the Gaussian Mixture PHD Filter2008Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    A recently established method for multi-target tracking which both estimates the time-varying number of targets and their states from a sequence of observation sets in the presence of data association uncertainty, detection uncertainty, noise and false alarms is the probability hypothesis density (PHD) recursion. The approach involves modeling the respective collections of targets and measurements as random finite sets and to propagate the posterior intensity, which is a first order statistic of the random finite set of targets, in time. A closed form solution to the PHD filter recursion for multi-target tracking is provided by the Gaussian Mixture Probability Hypothesis Density filter (GM-PHD filter), whose posterior intensity function is estimated by a sum of weighted Gaussian components, including means, weights and covariances that can be propagated analytically in time. Besides the GM-PHD filter algorithm implementation, choose the probability density function for representing target births in GM-PHD recursion and true target trajectory generation to get best tracking performance is a challenge and is the purpose of this thesis work. One reference to judge the performance of the algorithm is the target detection time, as given in this thesis.

  • 158. Chen, Yuanfang
    et al.
    Li, Mingchu
    Shu, Lei
    Wang, L.
    Duong, Quang Trung
    Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation.
    The scheme of mitigating the asymmetric links problem in wireless sensor networks2010Konferensbidrag (Refereegranskat)
    Abstract [en]

    This paper investigates the radio irregularity (RI) phenomenon and its impact on communication performance in wireless sensor networks (WSNs). Based on the theoretical analysis, we find that the RI phenomenon induces the asymmetric links problem. According to this discovery, we propose a novel HCT (Hop-count Correction Tree) scheme to handle this problem. HCT utilizes graph theoretical method to get a search tree and correct the hop count error that appears in the adjacent nodes. Practical results obtained from testbed experiments demonstrate that this solution can greatly improve localization accuracy in the presence of RI

  • 159.
    Cherukula, Pardhasaradhi Reddy
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap.
    Close talk Speech enhancement in Linear Microphone array for Laptop application2012Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    Now a day’s communication through laptops is drastically increasing in numerous fields. During the communication between person to person through laptops the speech signals is contaminated by the other speech interference signals. In order to enhance the desired speech signal from the noisy environment there are many algorithms are proposed in speech signal processing. This thesis work studies about the suppression of interference signals produced by the surrounding environments for the close talk applications of laptop. In this thesis work uses the three microphones of linearly equi spaced in 3D-co-ordinate system. The speech enhancement algorithms implemented in microphone array were wiener beamforming, Generalized sidelobe canceller using LMS and N-LMS. In order to enhance the desired speech signal with good quality, compares the result of each algorithm using quality metrics like SNR, SNRI and PESQ. The implementation and validation of the algorithms is simulated in Matlab. The quality metrics taken is SNRI and PESQ. In PESQ the output signal is compared with the original clean speech signal and gives the quality measure of the output signal. The SNR tests were conducted for the different input SNR values according to 0dB, 5dB, 10dB, 15dB, 20dB and 25dB. The Simulation result shows that the wiener beamformer effective noise suppression i.e SNRI is 27.9869dB and maintains the speech quality i.e PESQ measurement is 1.459.The effective noise suppression i.e SNRI of the GSC using LMS is 6.0206 dB higher than the wiener beamformer and speech quality is slightly incremental. Comparing the results of GSC using LMS and N-LMS algorithms, The GSC using N-LMS gives the effective noise suppression 3.48dB higher than the GSC using LMS and speech quality i.e PESQ is slightly decreases.

  • 160.
    CHETACHI, UMUNNA CHRISTIAN CHEZZ
    Blekinge Tekniska Högskola, Sektionen för teknik, Avdelningen för telekommunikationssystem.
    Security and Performance Analysis of Topology-Based Intrusion Detection System in Ad Hoc Networks2009Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [sv]

    Som Trådlöst ad-hoc-nätverk blir en allt mer framväxande teknik med en central plats i spännande forskningsområden som har lockat många forskning uppmärksamhet i kommunikationssystem, behovet av effektiva intrångslarm detekteringsmetoder att förbättra nät-säkerhet och integritet har fått betydande utrymme. För upptäckt av intrång system (IDS) i ad hoc-nät ger en effektiv metod för att förbättra säkerhet, energieffektivitet för avsändande noder som möjliggör bättre fysisk återanvändning av trådlös kanal och kontroll över nätverkets resurser för att optimera nod säkerhet och överföra makt därigenom bibehålla nätverksanslutning. Men dessa globala egenskaper har förfalskats på grund av brist på lämpliga Intrusion Detection System som leder till svår nätverk misslyckanden som är bane för nästa generation ad hoc-nät. I denna avhandling har vi undersöka några trådlösa ad hoc-säkerhet attacker och svagheter i förhållande till topologi kontrollera och utvärdera deras resultat under fientliga miljöer. Vi föreslår en ny Distributed Intrusion Detection System (DIDS) som innefattar regel-baserade kluster topologi betydelse för både trådlösa sensornätverk (WSNs) och Mobile ad hoc-nätverk (MANETs) för att fastställa deras säkerhet och prestanda i ansökan specifika miljöer. Vår DIDS drar slutsatser av intrång genom att jämföra avvikande mönster från paketkoppling spår av sända och ta emot signalen befogenheter, förhållandet paketkoppling ankomst priser och brist på radiomottagare paketkoppling makten tröskelvärden använder buffert fönster räkna. Därför utvärderar vi våra intrångslarm upptäckt mekanismen för en jammer attack och observera effekten på nätet genomströmning. Vår inställning är simuleras med hjälp av OPNET ® simulator. Simulering Resultaten visar att detekteringskapaciteten av våra system under en Denial of Service (DoS) (jammer) attack, ökar bit felfrekvenser, ökade överlämna dröjsmål svaren och betydande minskning i både signal-brus befogenheter och den genomsnittliga nät kapacitet på grund till förekomsten av jammer attack som ligger till grund för vår analys krävs för att upprätthålla effektiv energianvändning och förbättra säkerheten i ad hoc-nätverk.

  • 161.
    Chintakuntla, Abhiram
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap.
    Keystroke Resilience in Laptop Acoustics Using Wiener Beamformer and Spectral Subtraction2012Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    Nowadays, Laptops and Personal computers are increasingly being used to capture audio in various communication scenarios such as video conferencing, recordings of meetings, VOIP communications and lectures for archival purposes and audio/video instant messaging. Many of these scenarios often face unique problem of additive noise that of the user simultaneously typing on the keyboard. For example, to take notes while recording a meeting using the laptops local microphone array. The quality of speech signals captured by microphones is severely distorted by Keystroke noise which degrades the quality of speech. These degradation's lower the intelligibility of speech signal, So that listener’s ability to understand the message suffers. Thus there is a need to suppress the keystroke noise in recorded speech signals that results in significant perceptual improvement. In this thesis, the enhancement of speech signal by the suppression of keystroke noise is done through a unique combination of two algorithms i.e. Wiener Beamformer and Spectral Subtraction by Geometrical Approach. Wiener Beamformer is implemented based on subband approach. Both the algorithms individually and unique combination of both are implemented and performance is compared relative to each other by considering different performance measures i.e. Signal to Noise ratio (SNR) Improvement and PESQ. All the systems are tested with various positions of noise and speech source with reference to the microphone array and distance from speech source to microphone array is also varied. Analysing the results, the unique combination of both the algorithms proved to be efficient in terms of SNR improvement when compared to individual systems. The Spectral Subtraction algorithm does not improve the speech intelligibility effectively whereas Wiener Beamformer as an individual system works perfectly with keystroke noise suppression in speech signal and have relatively high PESQ values when compared to the values of combined system. Finally, the Spectral Subtraction reduces PESQ significantly at low SNR, the Spectral Subtraction adds few dB in improvement when combined with Weiner Beamformer, but at very high cost of decrement of PESQ.

  • 162.
    Chittajallu, Sai Kiran
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap.
    MOBILE PHONE ACOUSTICS: PERFORMANCE EVALUATION OF SPECTRAL SUBTRACTION AND ELKO’S ALGORITHM USING SPEECH QUALITY METRICS2013Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    In recent years a great deal of effort has been expended to develop methods that determine the quality of speech through the use of comparative algorithms. These methods are designed to calculate an index value of quality that correlates to a mean opinion score given by human subjects in evaluation sessions. In this work PESQ (ITU-T Recommendation P.862) which is the new ITU-T benchmarking for objective measurement of speech quality. In mobile phone acoustics, the presence of noise and room reverberation play a vital role in degrading the speech signal and therefore, spectral subtraction and Elko’s beamformer has been used for noise reduction. Weighted Overlap and Add method (WOLA) filter bank is used for frequency domain analysis of the speech signal. Elko’s algorithm is used for designing a differential microphone array, implemented by connecting two omni directional elements to form back-to-back cardioid directional microphones. The output from the Elko’s beamformer is then used as an input to spectral subtraction based on minimum statistics reducing further noise and to enhance the quality of the speech signal. The performance of this system is analysed by calculating the value of PESQ as a speech quality measure. The better the value the PESQ, the better is the output speech quality. Signal to Noise Ratio (SNR) is used to measure the amount of noise in the restored speech signal. Reverberation Index is also used to measure the amount of reverberation effect present in the restored speech signal.

  • 163.
    CHODISETTI, RAKESH ANIL
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    DASARI, BALA RAMA KRISHNA
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    H.264 VIDEO CODING ARTIFACTS: MEASUREMENT AND REDUCTION OF FLICKERING2014Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    H.264 video compression technique is currently one of the most widely used video codec for video compression and transmission. It is very important to maintain the quality of video with compression. Our research work mainly focuses on finding the various artifacts caused in the video due to compression using H.264 video technique. Blocking, ringing and flickering are the main artifacts found in the H.264 compression. Blocking and ringing are reduced mostly with the use of in loop filter in the H.264 codec. Even though blocking and ringing are reduced successfully, flickering is still seen in intra coded frames. Flickering is a temporal artifact which is otherwise known as mosquito noise is caused by change in the luminance values of the stationary region due to use of various prediction techniques . We propose a temporal median filter which can successfully reduce flickering effect in the H.264 video. The performance of the proposed filter is evaluated using sum of squared difference (SSD) metric.

  • 164.
    Chokkarapu, Anil
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap.
    Acoustic Fan Noise Cancellation in a Laptop Microphone System2012Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    Speech communication involving audio conferencing, video conferencing, teleconferencing via laptops became greatly influenced in office environments i.e. between employer and employee, and also influenced in personal life meetings between friends or in-between parents and children. These meeting conversations will mostly disturbs by annoying noise, i.e. fan noise which is produced by laptop cooling fan, which suffers at the both ends of communication due to this noise. With this noise effect the intelligibility of original speech is degraded between the conversations of meetings. So there is need of enhancing the speech from noisy speech environment in the communication. Thus speech enhancement is emerging technology in the communication and signal processing filed. So this thesis focuses on attenuating the noise produced by laptop cooling fan, with use of different speech enhancement algorithms. In this thesis we implement a multichannel Microphone Array (MA) of linearly arranged two microphones with different speech enhancement algorithms in spatial frequency domain. As the implementation involves frequency domain, we design a filter bank which is back bone structure of thesis, which is used to transform the signals received by microphone array into subband sequence, this subbands are processed through enhancement algorithms to attenuate noise, and then finally used to reconstruct the estimated original speech signal in time domain. The speech enhancement algorithms, involve beamforming technique i.e. Wiener BeamForming (WBF), and Spectral Subtraction (SS). Here we utilize a Direction Of Arrival (DOA) technique, to localize the speech source based on Time Difference Of Arrival (TDOA) in frequency domain only. Here we implements different systems involving individual WBF and SS algorithms, and also hybrid combination of algorithms WBF and SS, to suppress the fan noise. These systems were implemented at different positions of speech and noise sources. These systems were implemented and evaluated using simulation tool Matlab. The objective quality measures used to validate the systems are Signal to Noise Ratio Improvement (SNRI) and Perceptual Evaluation Speech Quality (PESQ) measure. The systems were validated with a pure speech combination of male and female sampled at 16 KHz, and fan noise recorded in the real time of anechoic environment. The systems are simulated at different SNR ratios of 0dB, 5dB, 10dB, 15dB, 20dB. The simulation result shows that hybrid system proves to be efficient in reduction of noise at higher SNR ratios, with compromise of speech quality. Whereas the individual beamformer system proves to be very highly efficient in reducing the noise while maintaining high quality in speech at both high and low SNR’s, whereas the spectral subtraction individual alone reduces a noise sufficiently only but its speech quality is very poor in performances. Also at different positions the systems were simulated, at the position where DOA of noise is 90 then all systems works highly efficient in reducing noise since SNRI are around 40dB to 50 dB while maintaining speech quality.

  • 165.
    Chowdary, Gannamaneni Geetha
    Blekinge Tekniska Högskola, Sektionen för ingenjörsvetenskap.
    Acoustic Echo cancellation inside a Conference Room using Adaptive Algorithms2012Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    The whole dimension of communications has been changed by the rapid growth of technology. Today people are more interested in hands-free com-mucation, which makes use of loud speaker and high gain microphone, in place of the old modeled wired telephone. The main advantage of wireless system is that, more than one person can participate in conversation while freely moving in the room. The presence of large acoustic coupling between speaker and microphone would produce a loud acoustic echo making the con-versation difficult. The term Acoustic Echo Cancellation (AEC) refers to a process of removing echo from the received signal that contains one or more delayed signals (copies of the original signal). The primary step while cancelling an echo is to identify the transmitted signal which reappears with some delay. Once the echo is identified it is cancelled by subtracting from transmitted signal. Echo cancellation can be done using either echo suppressors or echo cancellers, or in some case both. But suppressors support only half duplex communication leading to the invention of echo cancellers which allows both the speakers to talk at the same time. The main objective of this research is to model a room and cancel the acoustic echo being generated by a speaker and microphone. This dissertation provides a comparison of LMS, NLMS, LLMS and RLS adaptive algorithms in terms of echo return loss enhancement ( ERLE) value and provides the best suitable algorithm for usage in adaptive filters for AEC. AEC is simulated and results are evaluated by using Matlab

  • 166.
    Chowdhury, Moyamer
    et al.
    Blekinge Tekniska Högskola, Sektionen för teknik, Avdelningen för telekommunikationssystem.
    Alam, Aminul
    Blekinge Tekniska Högskola, Sektionen för teknik, Avdelningen för telekommunikationssystem.
    Study Comparison of WCDMA and OFDM2007Självständigt arbete på avancerad nivå (magisterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    Wideband Code Division Multiple Access (WCDMA) is one of the main technologies for the implementation of third-generation (3G) cellular systems. It is based on radio access technique proposed by ETSI Alpha group and the specifications was finalised 1999. WCDMA is also known as UMTS and has been adopted as a standard by the ITU under the name “IMT-2000 direct spread”. The implementation of WCDMA will be a technical challenge because of its complexity and versatility. The complexity of WCDMA systems can be viewed from different angles: the complexity of each single algorithm, the complexity of the overall system and the computational complexity of a receiver. In WCDMA interface different users can simultaneously transmit at different data rates and data rates can even vary in time. WCDMA increases data transmission rates in GSM systems by using the CDMA air interface instead of TDMA. WCDMA is based on CDMA and is the technology used in UMTS. WCDMA is the dominating 3G technology, providing higher capacity for voice and data and higher data rates. The gradual evolution from today's systems is driven by demand for capacity, which is required by new and faster data based mobile services. WCDMA enables better use of available spectrum and more cost-efficient network solutions. The operator can gradually evolve from GSM to WCDMA, protecting investments by re-using the GSM core network and 2G/2.5G services. Orthogonal Frequency Division Multiplexing (OFDM) - technique for increasing the amount of information that can be carried over a wireless network uses an FDM modulation technique for transmitting large amounts of digital data over a radio wave. OFDM works by splitting the radio signal into multiple smaller sub-signals that are then transmitted simultaneously at different frequencies to the receiver. OFDM reduces the amount of crosstalk in signal transmissions. 802.11a WLAN, 802.16 and WiMAX technologies use OFDM. It's also used in the ETSI's HiperLAN/2 standard. In addition, Japan's Mobile Multimedia Access Communications (MMAC) WLAN broadband mobile technology uses OFDM. In frequency-division multiplexing, multiple signals, or carriers, are sent simultaneously over different frequencies between two points. However, FDM has an inherent problem: Wireless signals can travel multiple paths from transmitter to receiver (by bouncing off buildings, mountains and even passing airplanes); receivers can have trouble sorting all the resulting data out. Orthogonal FDM deals with this multipath problem by splitting carriers into smaller subcarriers, and then broadcasting those simultaneously. This reduces multipath distortion and reduces RF interference allowing for greater throughput. In this paper we have discussed about these two methods of third generation radio transmission system which are WCDMA and OFDM with various aspects. In between these two radio transmission technique, a better choice will be investigated.

  • 167.
    Chunduri, Krishna Chaitanya
    et al.
    Blekinge Tekniska Högskola, Sektionen för teknik, Avdelningen för signalbehandling.
    Gutti, Chalapathi
    Blekinge Tekniska Högskola, Sektionen för teknik, Avdelningen för signalbehandling.
    Implementation of Adaptive Filter Structures on a Fixed Point Signal Processor for Acoustical Noise Reduction2005Självständigt arbete på avancerad nivå (magisterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    The problem of controlling the noise level in the environment has been the focus of a tremendous amount of research over the years. Active Noise Cancellation (ANC) is one such approach that has been proposed for reduction of steady state noise. ANC refers to an electromechanical or electro acoustic technique of canceling an acoustic disturbance to yield a quieter environment. The basic principle of ANC is to introduce a canceling “anti-noise” signal that has the same amplitude but the exact opposite phase, thus resulting in an attenuated residual noise signal. Wideband ANC systems often involve adaptive filter lengths, with hundreds of taps. Using sub band processing can considerably reduce the length of the adaptive filter. This thesis presents Filtered-X Least Mean Squares (FXLMS) algorithm to implement it on a fixed point digital signal processor (DSP), ADUC7026 micro controller from Analog devices. Results show that the implementation in fixed point matches the performance of a floating point implementation.

  • 168. Claesson, Ingvar
    Dual Microphone input for Mobile Telephony2000Konferensbidrag (Refereegranskat)
  • 169. Claesson, Ingvar
    FFT i ett historiskt perspektiv1997Konferensbidrag (Refereegranskat)
    Abstract [en]

    Några nedslag i Fouriertransformens historia och FFT:ns ursprung från 1805 och framåt behandlas i denna populära betraktelse. Vi stannar upp ett slag hos Gauss, funderar över vad han gjort för oss ingenjörer och vi avslutar med en del praktiska konsekvenser och tips som är aktuella idag.

  • 170. Claesson, Ingvar
    FFT i ett historiskt perspektiv1997Konferensbidrag (Refereegranskat)
  • 171. Claesson, Ingvar
    FFT i ett historiskt perspektiv1997Ingår i: SVIB vibrationsnytt , ISSN 0281-6830, Vol. 14/15, nr 1, s. 11-15Artikel i tidskrift (Refereegranskat)
  • 172. Claesson, Ingvar
    et al.
    Dahl, Mattias
    Nordebo, Sven
    Applied Complex Chebyshev Optimization Using Dual Nested Complex Approximation2001Konferensbidrag (Refereegranskat)
  • 173. Claesson, Ingvar
    et al.
    Dahl, Mattias
    Nordebo, Sven
    Nordholm, Sven
    Acoustic Echo Cancelling with microphone arrays1995Konferensbidrag (Refereegranskat)
  • 174. Claesson, Ingvar
    et al.
    Dahl, Mattias
    Nordebo, Sven
    Nordholm, Sven
    Chebyshev Optimization of Circular Arrays Inequalities1998Konferensbidrag (Refereegranskat)
  • 175. Claesson, Ingvar
    et al.
    Håkansson, Lars
    Active Control of Machine-Tool Vibration in a Lathe1997Konferensbidrag (Refereegranskat)
    Abstract [en]

    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.

  • 176. Claesson, Ingvar
    et al.
    Håkansson, Lars
    Adaptive Active Control of Machine-Tool Vibration In a Lathe1998Ingår i: International Journal of Acoustics and Vibration, ISSN 1027-5851, Vol. 3, nr 4, s. 155-162Artikel i tidskrift (Refereegranskat)
    Abstract [en]

    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.

  • 177.
    Claesson, Ingvar
    et al.
    Blekinge Tekniska Högskola, Institutionen för signalbehandling.
    Håkansson, Lars
    Blekinge Tekniska Högskola, Institutionen för signalbehandling.
    Lagö, Thomas L
    Robust Control of Machine-Tool Vibration in a Lathe1999Konferensbidrag (Refereegranskat)
    Abstract [en]

    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.

  • 178. Claesson, Ingvar
    et al.
    Nilsson, Andreas
    Cancellation of Humming GSM Mobile Telephone Noise2003Konferensbidrag (Refereegranskat)
    Abstract [en]

    A sometimes annoying problem in the most internationally widespread cellular telephone system, the GSM system, is an interfering signal generated by the switching nature of TDMA cellular telephone system. A humming noise originating from the speech frames, equivalent to 160 samples of data corresponding to 20 ms at 8 kHz sampling rate is sometimes clearly audible. This paper describes a study of two different software solutions designed to suppress such interference internally in the mobile handset. The methods are Notch Filtering, which is performed on a sample-per-sample basis, and Speech Frame Noise Cancellation, which is an alternative method employing correlators and subtraction, similar to Active Noise Control.

  • 179. Claesson, Ingvar
    et al.
    Nilsson, Andreas
    GSM TDMA Frame Rate Internal Active Noise Cancellation2003Ingår i: International Journal of Acoustics and Vibration, ISSN 1027-5851, Vol. 8, nr 3, s. 159-166Artikel i tidskrift (Refereegranskat)
    Abstract [en]

    A common problem in the world's most widely-used cellular telephone system, the GSM system, is the interfering signal generated by the switching nature of TDMA cellular telephony in handheld and other terminals. Signals are sent as chunks of data, speech frames, equivalent to 160 samples of data corresponding to 20 ms at sampling rate of 8 kHz. This paper describes a study of two different software solutions designed to suppress such interference internally in the mobile handset. The methods are 1) notch filtering, which is multiplicative in frequency, and 2) subtractive noise cancellation, which is an alternative method employing correlators. The latter solution is a straigtforward, although somewhat unorthodox, application of "in-wire" active noise control. Since subtraction is performed directly in the time domain, and we have access to the state of the mobile, it is also possible to consider a recurring pause in the interference caused by the idle frame in the transmission, when the mobile listens to other base stations communicating. More complex control algorithms, based on the state of the communication between the handset and the base station, can be utilised.

  • 180. Claesson, Ingvar
    et al.
    Nordholm, Sven
    A Spatial Filtering Approach to Robust Adaptive Beamforming1992Ingår i: IEEE Transactions on Antennas and Propagation, ISSN 0018-926X , Vol. 40, nr 9, s. 1093-1096Artikel i tidskrift (Refereegranskat)
    Abstract [en]

    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.

  • 181. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Robust Adaptive Beamforming using Spatial Filter Design1990Rapport (Övrigt vetenskapligt)
  • 182. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Bengtsson, Bengt
    Car Perfomance of an Adaptive Microphone Array1992Konferensbidrag (Refereegranskat)
  • 183. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Bengtsson, Bengt
    Car Performance of an Adaptive Microphone Array1992Konferensbidrag (Refereegranskat)
    Abstract [en]

    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

  • 184. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Bengtsson, Bengt
    Car Performance of an Adaptive Microphone Array1992Konferensbidrag (Refereegranskat)
    Abstract [en]

    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.

  • 185. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Bengtsson, Bengt
    Car Performance of an Adaptive Microphone Array1992Konferensbidrag (Refereegranskat)
  • 186. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Bengtsson, Bengt
    Eriksson, Per
    A Multi-DSP implementation of a Broadband Adaptive Beamformer for use in a Hands-free Mobile Radio Telephone1991Ingår i: IEEE Transactions on Vehicular Technology, ISSN 0018-9545 , Vol. 40, nr 1, s. 194-202Artikel i tidskrift (Refereegranskat)
    Abstract [en]

    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.

  • 187. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Eriksson, Per
    Noise Canceling Convergence Rates for the LMS Algorithm1991Ingår i: Mechanical systems and signal processing, ISSN 0888-3270, E-ISSN 1096-1216, Vol. 5, nr 5, s. 375-388Artikel i tidskrift (Refereegranskat)
  • 188. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Eriksson, Per
    Noise Cancelling Convergence Rates for the LMS Algorithm1991Ingår i: Mechanical Systems & Signal Processing, ISSN 0888-3270 , Vol. 5, nr 5, s. 375-388Artikel i tidskrift (Refereegranskat)
  • 189. Claesson, Ingvar
    et al.
    Nordholm, Sven
    Eriksson, Per
    Wiener Solution for the Broadband Griffiths-Jim Beamformers1990Konferensbidrag (Refereegranskat)
    Abstract [en]

    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.

  • 190. Claesson, Ingvar
    et al.
    Rossholm, Andreas
    Notch Filtering of Humming GSM Mobile Telephone Noise2005Konferensbidrag (Refereegranskat)
    Abstract [en]

    A common problem in the world's most widespread cellular telephone system, the GSM system, is the interfering signal generated in TDMA cellular telephony. The infamous "bumblebee" is generated by the switching nature of TDMA cellular telephony, the radio circuits are switched on and off at a rate of approximately 217 Hz (GSM). This paper describes a study of two solutions for eliminating the humming noise with IIR notch filters. The simpler one is suitable for any exterior equipment. This method still suffers from a small residual of the noise, resulting from the IDLE slots of the sending mobile. The more advanced IIR structure for use within the mobile also eliminates this residual.

  • 191.
    Claesson, Lena
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    Remote Electronic and Acoustic Laboratories in Upper Secondary Schools2014Licentiatavhandling, sammanläggning (Övrigt vetenskapligt)
    Abstract [en]

    During a substantial part of their time young people of today actually live in a virtual world. The medial evolution has also influenced education and today much research work basically concerns the transfer of the physical world into the virtual one. One example is laboratories in physical science that are available in virtual rooms. They enable studentsto sit at home in front of a computer and on screen watch and operate the physical equipment in the laboratory at school. It is a general agreement that laboratory lessons are necessary in subjects such as physics, chemistry and biology. Physical experiments provide a great way for students to learn more about nature and its possibilities as well as limitations. Experimental work can be provided bylaboratories in three different categories; 1) hands-on, 2) remote and 3) simulated. This thesis concerns the usage of remotely controlled laboratories in physics education at an upper secondary school. It is based on work carried out in a joint project between Katedralskolan (upper secondary school), Lund, Sweden, and Blekinge Institute of Technology (BTH). The object with this project is to investigate feasibility of using the VISIR (Virtual Instruments System in Reality) technology for remotely controlled laboratories, developed at BTH, in upper secondary schools. This thesis consists of an introduction, followed by three parts where the first part concerns the introduction of the remote lab to students and the usage of the remote lab by students at the upper secondary school, Katedralskolan. Both first year students and third year students carried out experiments using the remote lab. The second part concerns activities carried out by 2 teachers and 94 students using the remote laboratory VISIR. An integration of VISIR with the learning management system used at school is described. Teaching activities carried out by teachers at Katedralskolan involving the VISIR lab are discussed, e.g., an exam including problems of experimental work using the VISIR lab and an example of a student report. Survey results on student satisfaction with the VISIR lab at BTH and the perception of it are presented, indicating that VISIR is a good learning tool. Furthermore, the survey resulted in a proposal of improvements in the VISIR lab user interface. Finally, the third part focuses on enhancements of the VISIR lab at BTH. An improved version in the VISIR user interface is presented. New iPad and smart phone availability of the VISIR lab is presented. Electronic experiments for upper secondary school students are described in detail and examples of suitable configurations are given. A new VISIR acoustic lab has beenimplemented and initial experimentation by upper secondary school students have been carried out. The outcomes from these experiments are discussed.

  • 192.
    Claesson, Lena
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    Håkansson, Lars
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    Displacement measurements versus time using a remote inclined plane laboratory2016Ingår i: Proceedings of 2016 13th International Conference on Remote Engineering and Virtual Instrumentation, REV 2016, IEEE Press, 2016, s. 355-356Konferensbidrag (Refereegranskat)
    Abstract [en]

    This paper describes a remote implementation of Galileos inclined plane experiment, focused on secondary school students. A remotely controlled inclined plane has been designed and implemented in the VISIR lab at Blekinge Institute of Technology (BTH), Sweden. In this demo session, it will be demonstrated how to perform measurements remotely in the remotely controlled Inclined Plane Laboratory. A web camera is used to show the experiment. Data concerning the distance a cube has slided down the inclined plane are collected. These data are stored in a file and can subsequently be analyzed by the students. The friction acting on the cube sliding down the inclined plane and its acceleration may for instance be investigated.

  • 193.
    Claesson, Lena
    et al.
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    Lundberg, Jenny
    Linnéuniversitetet, SWE.
    Zackrisson, Johan
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    Johansson, Sven
    Blekinge Tekniska Högskola, Fakulteten för teknikvetenskaper, Institutionen för tillämpad signalbehandling.
    Hakansson, Lars
    Linnéuniversitetet, SWE.
    Expert Competence in Remote Diagnostics - Industrial Interests, Educational Goals, Flipped Classroom & Laboratory Settings2018Ingår i: ONLINE ENGINEERING & INTERNET OF THINGS / [ed] Auer, ME Zutin, DG, 2018, s. 438-451Konferensbidrag (Refereegranskat)
    Abstract [en]

    The manufacturing industry are dependent of engineering expertise. Currently the ability to supply the industry with engineering graduates and staff that have an up-to-date and relevant competences might be considered as a challenge for the society. In this paper an education approach is presented where academia - industry - research institutes cooperate around the development and implementation of master level courses. The methods applied to reach the educational goals, concerning expert competence within remote diagnostics, have been on site and remote lectures given by engineering, medical and metrology experts. The pedagogical approach utilized has been flipped classroom. The main results show that academic courses developed in cooperation with industry requires flexibility, time and effort from the involved partners. The evaluation interviews indicate that student are satisfied with the courses and pedagogical approach but suggests more reconciliation meetings for course development. Labs early in the course was considered good, and division of labs at the system and the component level. However further long-term studies of evaluation of impact is necessary.

  • 194. Cornelius, Per
    Subband Beamforming for Speech Enhancement within a Motorcycle helmet2005Licentiatavhandling, sammanläggning (Övrigt vetenskapligt)
    Abstract [en]

    The increased mobility in society has led to a need for convenient mobile communication in many different type of environments. Environments such as a motorcycle helmet, engine rooms and most industrial sites share a common challenge in that they often offer significant acoustic background noise. Noise reduces the speech intelligibility and consequently limits the potential of mobile speech communications. Existing single channel solutions for speech enhancement may perform adequately when the level of noise is moderate. When the noise level becomes significant, additional use of the spatial domain in order to successfully perform speech enhancement is a potential solution. This is achieved by including several microphones in an array placed in the vicinity of the person speaking. A beamforming algorithm is hereby used to combine the microphones such that the desired speech signal is enhanced. The interest in using microphone arrays for broadband speech and audio processing has increased in recent years. There have been a considerable amount of interesting applications published using beamforming techniques for hands-free voice communication in cars, hearing-aids, teleconferencing and multimedia applications. Most of proposed solutions deal exclusively with environments where the noise is moderate. This thesis is a study of noise reduction in a helmet communication system on a moving motorcycle. The environment is analyzed under different driving conditions and a speech enhancement solution is proposed that operates successfully in all driving conditions. The motorcycle environment can exhibit extremely high noise levels, when driving at high speed, while it can produce a low noise levels at moderate speeds. This fact implies that different solutions are required. It is demonstrated in the thesis that a cascaded combination of a calibrated subband beamforming technique, together with a single channel solution provides good results at all noise levels. The proposed solution operates in the frequency domain, where all microphone signals are decomposed by a subband filter bank prior to the speech enhancement processing. Since the subband transformation is an important component of the overall system performance, a method for filter bank design is also provided in the thesis. The design is such that the aliasing effects in the transformations are minimized while a small delay of the total system is maintained.

  • 195. Cornelius, Per
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Microphone array system for speech enhancement in a motorcycle helmet2005Rapport (Övrigt vetenskapligt)
    Abstract [en]

    In this report a real case study of the sound environment within a helmet while driving motorcycle is investigated. A solution to perform speech enhancement is proposed for the purpose of mobile speech communication. A microphone array, mounted onto the face shield in front of the user's mouth, is used to capture the spatio-temporal properties of the acoustic wave ¯eld inside the helmet. The power of the spatially spread noise within the helmet is small when standing still while it may heavily exceed the power of the speech when driving at high speeds. This will result in dramatically reduced speech intelligibility in the communication channel. The highly dynamic noise level imposes a challenge for existing speech enhancement solutions. We propose a subband adaptive system for speech enhancement which consists of a soft constrained beamformer in cascade with a signal-to-noise ratio dependent single microphone solution. The beamformer make use of a calibration signal gathered in the actual environment from the speaker's position. This calibration procedure e±ciently captures the acoustical properties in the environment. Evaluation of the beamformer and the single microphone algorithm, both as either parts by them selves and as a cascaded structure, together with the optimal subband Wiener solution is presented. It is shown that a cascaded combination of the calibrated subband beamforming technique together with the single channel solution outperforms either one by it self, and provides near optimal results at all noise levels.

  • 196. Cornelius, Per
    et al.
    Yermeche, Zohra
    Grbic, Nedelko
    Claesson, Ingvar
    A Spatially Constrained Subband Beamforming Algorithm for speech enhancement2004Konferensbidrag (Refereegranskat)
    Abstract [en]

    This paper discusses speech enhancement in an enclosed environment such as communication in a motorcycle helmet. A new constrained subband adaptive beamformer is proposed, which uses the concept of an earlier proposed calibrated beamformer mainly developed for a hands-free in-car environment. The highly non-stationary nature of the disturbing sound field encountered in an motorcycle helmet and the fact that the source is situated in the extreme nearfield of the array, causes the beamformer to produce an unwanted fluctuation in the output level. The spatially constrained beamformer proposed in this paper makes sure that the output maintains a constant gain, as long as the corresponding source originates from the desired location.

  • 197. Cresp, Gregory
    et al.
    Dam, Hai Huyen
    Zepernick, Hans-Jürgen
    Design of Sequence Family Subsets Using a Branch and Bound Technique2009Ingår i: IEEE Transactions on Information Theory, ISSN 0018-9448, E-ISSN 1557-9654, Vol. 55, nr 8, s. 3847-3857Artikel i tidskrift (Refereegranskat)
    Abstract [en]

    The number of spreading sequences required for Direct Sequence Code Division Multiple Access (DS-CDMA) systems depends on the number of simultaneous users in the system. Often a sequence family provides more sequences than are required; in many cases the selection of the employed sequences is a computationally intensive task. This selection is a key consideration, as the properties of the sequences assigned affect the error performance in the system. In this paper, a branch and bound algorithm is presented to perform this selection based on two different cost functions. Numerical results are presented to demonstrate the improved performance of this algorithm over previous work.

  • 198.
    Czynszak, Szymon
    Blekinge Tekniska Högskola, Sektionen för datavetenskap och kommunikation.
    Decoding algorithms of Reed-Solomon code2011Självständigt arbete på avancerad nivå (masterexamen)Studentuppsats (Examensarbete)
    Abstract [en]

    Reed-Solomon code is nowadays broadly used in many fields of data transmission. Using of error correction codes is divided into two main operations: information coding before sending information into communication channel and decoding received information at the other side. There are vast of decoding algorithms of Reed-Solomon codes, which have specific features. There is needed knowledge of features of algorithms to choose correct algorithm which satisfies requirements of system. There are evaluated cyclic decoding algorithm, Peterson-Gorenstein-Zierler algorithm, Berlekamp-Massey algorithm, Sugiyama algorithm with erasures and without erasures and Guruswami-Sudan algorithm. There was done implementation of algorithms in software and in hardware. Simulation of implemented algorithms was performed. Algorithms were evaluated and there were proposed methods to improve their work.

  • 199. Dahl, Mattias
    Acoustic Noise and Echo Cancelling: Microphone Array Methods and Applications1997Licentiatavhandling, sammanläggning (Övrigt vetenskapligt)
    Abstract [en]

    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.

  • 200. Dahl, Mattias
    Acoustic Noise and Echo Cancelling: Microphone Array Methods and Applications1997Rapport (Övrigt vetenskapligt)
    Abstract [en]

    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.

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