Change search
Refine search result
123456 51 - 100 of 279
CiteExportLink to result list
Permanent link
Cite
Citation style
  • apa
  • harvard1
  • ieee
  • modern-language-association-8th-edition
  • vancouver
  • Other style
More styles
Language
  • de-DE
  • en-GB
  • en-US
  • fi-FI
  • nn-NO
  • nn-NB
  • sv-SE
  • Other locale
More languages
Output format
  • html
  • text
  • asciidoc
  • rtf
Rows per page
  • 5
  • 10
  • 20
  • 50
  • 100
  • 250
Sort
  • Standard (Relevance)
  • Author A-Ö
  • Author Ö-A
  • Title A-Ö
  • Title Ö-A
  • Publication type A-Ö
  • Publication type Ö-A
  • Issued (Oldest first)
  • Issued (Newest first)
  • Created (Oldest first)
  • Created (Newest first)
  • Last updated (Oldest first)
  • Last updated (Newest first)
  • Disputation date (earliest first)
  • Disputation date (latest first)
  • Standard (Relevance)
  • Author A-Ö
  • Author Ö-A
  • Title A-Ö
  • Title Ö-A
  • Publication type A-Ö
  • Publication type Ö-A
  • Issued (Oldest first)
  • Issued (Newest first)
  • Created (Oldest first)
  • Created (Newest first)
  • Last updated (Oldest first)
  • Last updated (Newest first)
  • Disputation date (earliest first)
  • Disputation date (latest first)
Select
The maximal number of hits you can export is 250. When you want to export more records please use the Create feeds function.
  • 51. Fiedler, Markus
    et al.
    Hasslinger, Gerhard
    Waiting Time Quantiles for the Gaussian Voice Traffic Model2002Other (Other academic)
    Abstract [en]

    Traffic models with a rate varying according to a Gaussian distribution are commonly used to evaluate statistical multiplexing in telecommunication systems. The superposition of a large number of homogeneous Markovian On-Off sources asymptotically approaches an Ornstein-Uhlenbeck process (OUP) which represents a Gaussian process with exponential autocorrelation function. The workload analysis for a multiplexer with OUP input leads to a simple and clearly structured dependence of waiting time quantiles on system and quality of service parameters. Semi-Markovian fitting procedures as well as the fluid flow method provide independent analysis ap-proaches at an assured accuracy level. The comparison of the corresponding waiting time quantiles shows the suitability of assuming traffic to be Gaussian already for a moderate number of traffic sources.

  • 52. Fiedler, Markus
    et al.
    Krieger, Udo R.
    The impact of varying channel capacity on the quality of advanced data services in PCS networks2000Conference paper (Refereed)
    Abstract [en]

    We develop a unifying framework to perform the end-to-end quality of service (QoS) management of advanced data services in PCS networks. For this purpose, we first describe a generic fluid-flow model with Markov-modulated input flows and variable service rates depending on a Markovian environment for the wireless part of the transport path. Then we indicate how it can be used to evaluate the QoS of the data transport and to what extent the varying capacity of the transport channel influences the latter.

  • 53. Fiedler, Markus
    et al.
    Tutschku, Kurt
    Application of the stochastic fluid flow model for bottleneck identification and classification2003Conference paper (Refereed)
    Abstract [en]

    Network performance management is facing the challenge of provisioning advanced services with stringent delay and throughput requirements. For this reason, shortage of network capacity implying delay or loss, so-called bottlenecks, have to be identified and to be classified. The latter tasks imply the need for tractable analytical performance models. We identify the stochastic fluid flow model, which is based on bit rates and its statistics, as a possible candidate of being capable of describing qualitative behaviour of bottlenecks. In this work, we show how total and individual bit rate statistics at the output of a bottleneck are calculated via the stochastic fluid flow model. From this, we deduce some general behaviours and classification criteria for bottlenecks.

  • 54. Fiedler, Markus
    et al.
    Tutschku, Kurt
    Carlsson, Patrik
    Nilsson, Arne A.
    Identification of performance degradation in IP networks using throughput statistics2003Conference paper (Refereed)
    Abstract [en]

    To be able to satisfy their users, interactive applications like video conferences require a certain Quality-of-Service from heterogeneous networks. This paper proposes the use of throughput histograms as Quality-of-Service indicator. These histograms are built from local, unsynchronized, passive measurements of packet streams from the viewpoint of an application. They can easily be exchanged between sender and receiver, and their comparison provides information about severity and type of a potential bottleneck. We demonstrate the usefulness of these indicators for evaluating the transport quality perceived by a video conferencing application and its users in the presence of a bottleneck.

  • 55. Fiedler, Markus
    et al.
    Voos, Holger
    New results on the numerical stability of the stochastic fluid flow model analysis2000Conference paper (Refereed)
    Abstract [en]

    The stochastic fluid flow model (SFF) is one of the leading models in performance evaluation for tele- and datacommunication systems, especially in fast packet-switching networks and ATM. However, the numerical analysis of the SFF is widely considered to be unstable. In this paper, some investigations and results are presented concerning the numerical stability of the SFF analysis also for large systems with finite buffer. We identify the main source of the numerical problems and give hints how to circumvent them. The usefulness of different solution methods are compared and the most robust methods for systems with large numbers of sources and large buffer sizes are identifed.

  • 56. Fundin, Per
    et al.
    Haan, Jan Mark de
    Grbic, Nedelko
    Claesson, Ingvar
    Nordholm, Sven
    Minimal Aliasing Subband System Identification2001Conference paper (Refereed)
    Abstract [en]

    Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. While the computational complexity is reduced, other undesired properties, such as signal delays and signal aliasing, are introduced. Aliasing effects may result in loss of perception in speech applications. A method for the design of oversampled filter banks is proposed to reduce these effects. The design method aims at reducing the inband aliasing as well as the reconstruction aliasing for the reason of achieving robustness when weighting in the subbands alters the subband signal phase and magnitude.

  • 57. Grbic, Nedelko
    Optimal and Adaptive Subband Beamforming2001Doctoral thesis, comprehensive summary (Other academic)
    Abstract [en]

    The increased use of personal communication devices, personal computers and wireless cellular telephones enables the development of new inter-personal communication systems. The merge between computers and telephony technologies brings up the demand for convenient hands-free communications. In such systems the users wish to lead a conversation in much the same way as in a normal person-to-person conversation. The advantages of hands-free telephones are safety, convenience and greater flexibility. In many countries and regions, hand held telephony in cars is prohibited by legislation. By placing the microphone far away from the user a number of disadvantages are introduced, which results in substantial speech distortion and poor sound quality. These disturbances are mainly caused by room reverberation and background noise. Furthermore, acoustic feedback generated at the near-end side is a problem for the far-end side talker, who will hear his/her own voice echoed with 100-200 ms delay, making speech conversation substantially more difficult. Digital filtering may be used to obtain a similar sound quality as for hand held telephony. Three major tasks must be addressed in order to improve the quality of hands-free communication systems; noise suppression, room reverberation suppression, and acoustic feedback cancellation of the hands-free loudspeaker. The filtering operation must perform the above mentioned tasks without causing severe near-end speech distortion. A properly designed broad-band microphone array is able to perform all the given tasks, i.e. speech enhancement, echo cancellation and reverberation suppression, in a concise and effective manner. This is due to the fact that the spatial domain may be utilized as well as the temporal domain. This thesis deals with the problem of specification and design of beamformers used to extract the source signal information. A new subband adaptive beamforming algorithm is proposed, where many of the drawbacks embedded in conventional adaptive beamforming are eliminated. Evaluation in a car hands-free situation show the benefits of the proposed method. Blind signal separation is discussed and a new structure based on frequency domain inverse channel identification and time domain separation, is proposed. Further, filter-bank properties and design are discussed together with performance limitations in subband beamforming structures.

  • 58. Grbic, Nedelko
    Performance Analysis and Bias Correction in Synchronized Acoustic 3D Measuring System2000Conference paper (Refereed)
  • 59. Grbic, Nedelko
    Speech Signal Extraction: A Multichannel Approach1999Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    In speech signal extraction the aim is to extract human speech in a physical environment by using microphones. In any real world environment there are many disturbance sources that cause unwanted sound pressure, which in turn may degrade the comprehension of the speech at the microphones. This licentiate thesis deals with the solution to this problem. Different approaches are proposed to solve the problem for many accounted scenarios in real life.

  • 60. Grbic, Nedelko
    et al.
    Dahl, Mattias
    Claesson, Ingvar
    Acoustic Echo Cancelling and Noise Suppression with Microphone Arrays1999Report (Other academic)
    Abstract [en]

    This report presents a method to achieve acoustic echo canceling and noise suppression using 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 specific case to the environment and the electronic equipment used. The method also continuously reduces environmental disturbances such as car engine noise and fan noise. The method is primarily aimed at hands free mobile telephones by suppressing the hands free loudspeaker and car cabin noise simultaneously. The report also contains an evaluation of the impact of echo and noise suppression on a real conversation, accomplished in a car using a microphone array.

  • 61.
    Grbic, Nedelko
    et al.
    Blekinge Institute of Technology, Department of Signal Processing.
    Haan, Jan Mark de
    Nordholm, Sven
    Blekinge Institute of Technology, Department of Signal Processing.
    Claesson, Ingvar
    Blekinge Institute of Technology, Department of Signal Processing.
    Design of Oversampled Uniform DFT Filter Banks with Reduced Inband Aliasing and Delay Constraints2001Conference paper (Refereed)
    Abstract [en]

    Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. Subband processing introduces transmission delays caused by the filter bank and signal degradations due to aliasing effects. One efficient way to reduce the aliasing effects is to allow a higher sample rate than critically needed in the subbands and thus reduce subband signal degradation. We suggest a design method, for a uniform DFT filter bank with any over sampling factor, where the total filter bank group delay may be specified, and where the aliasing and magnitude/phase distortions are minimized.

  • 62. Grbic, Nedelko
    et al.
    Nordberg, Jörgen
    Nordholm, Sven
    Subband Acoustic Echo Cancelling using LMS and RLS1999Report (Other academic)
    Abstract [en]

    The increasing use of modern hands free communication systems such as video conferencing, computer communications, and vehicle mounted cellular telephones brings the demand for high-quality acoustic echo cancellation up to focus. In these applications the echo path which has to be identified typically has long time duration, the order of 100 ms. For this identification the length of the filter will be long. This report evaluates the Normalized Least Mean Square (NLMS) and the Weighted Recursive Least Square (WRLS) algorithms for acoustic echo cancelling using a delayless subband scheme. Subband signal processing has shown to be efficient both when it comes to convergence rate and level of echo suppression. The evaluation is performed for real speech signals sampled from a conversation using a hands free set mounted in an automobile, and a conversation using conference telephony equipment in a conference room. A comparison of subband and fullband algorithms is made both with respect to the computational cost and level of echo suppression. Results show that when the impulse response is very long, i.e. in such environments as conference rooms, the subband approach is beneficial. In a car environment the size of enclosure and damping means that the response is quite short and a conventional echo canceller could perform as well as a subband echo canceller. In the study, finite word length effects have not been considered.

  • 63. Grbic, Nedelko
    et al.
    Nordholm, Sven
    Soft Constrained Subband Beamforming for Hands-Free Speech Enhancement2002Conference paper (Refereed)
    Abstract [en]

    This paper introduces a new constrained adaptive subband beamformer algorithm for speech enhancement in acoustic telecommunication systems. The solution relies on a pre-calculated source covariance matrix and recursive estimates of background noise- and handsfree signal covariance matrices. The constraint acts as an eye-opening in a vicinity of the near-field location of the source and degradations from steering-vector errors can therefor be made small. The algorithm is applied in subbands using a uniform multi channel over-sampled filterbank. Simulations with real speech recorded in an automobile hands-free environment show 19 dB noise reduction and 20 dB hands-free suppression.

  • 64. Grbic, Nedelko
    et al.
    Nordholm, Sven
    Cantoni, Antonio
    Optimal FIR Subband Beamforming for Speech Enhancement in Multipath Environments2003In: IEEE Signal Processing Letters, ISSN 1070-9908, E-ISSN 1558-2361, Vol. 10, no 11, p. 335-338Article in journal (Refereed)
    Abstract [en]

    This paper provides an analysis of optimal finite impulse response subband beamforming for speech enhancement in multipath environments. A modification of the direct path standard Wiener formulation is shown to give near-optimal performance when it comes to signal-to-noise plus interference ratio, by including coherent multipath propagation into the criterion.

  • 65.
    Grbic, Nedelko
    et al.
    Blekinge Institute of Technology, Department of Signal Processing.
    Nordholm, Sven
    Blekinge Institute of Technology, Department of Signal Processing.
    Johansson, Anders
    Blekinge Institute of Technology, Department of Signal Processing.
    Optimal Beamforming for Voice Input to Personal Communication Devices2000Conference paper (Refereed)
  • 66. Grbic, Nedelko
    et al.
    Nordholm, Sven
    Johansson, Anders
    Speech Enhancement for Hands-Free Terminals2001Conference paper (Refereed)
    Abstract [en]

    This paper discusses signal processing methods for speech extraction in use with voice communication applications such as personal digital assistants (PDA:s), mobile telephone terminals and personal computers. The user will be distant from the device and thus the speech signal entering the device will be subject to reverberation as well as disturbed by background noise. The proposed structure consists of a microphone array which allows for techniques of directional processing. Thus it will be able to enhance a desired speech source and suppress other speech sources in the background. Three different optimal beamforming methods are considered in a real world car hands-free environment; an optimal near-field array gain optimization procedure, a theoretical diffuse noise field model for a point source and a least squares solution.

  • 67. Grbic, Nedelko
    et al.
    Nordholm, Sven
    Nordberg, Jörgen
    Claesson, Ingvar
    A New Pilot-Signal based Space-Time Adaptive Algorithm2001Conference paper (Refereed)
    Abstract [en]

    In the application of adaptive antenna arrays to wireless communications, a known pilot signal sequence may be used for estimating the array response at the beginning of each data frame. This pilot sequence is usually very short and conventional training methods which estimate the array response, based solely on this training sequence, may incur large estimation errors. In this paper, we propose an online modified weighted recursive least squares type of training algorithm for estimating the optimal array response by exploiting information from the whole frame of the received signal. The benefits of the proposed algorithm is that tracking of coherent noise and interference signals is substantially improved, and the overall performance is increased. Simulation results show that the proposed method offers substantial improvement when compared to the conventional least squares method.

  • 68. Grbic, Nedelko
    et al.
    Tao, Xiao-Jiao
    Claesson, Ingvar
    Performance Improvement of Multiple Array 3D Sonic Digitizer System via Calibration1999Conference paper (Refereed)
    Abstract [en]

    Multiple array 3D sonic digitizer system is a product made for measuring the geometric structure of an arbitrary object in three dimension. This paper gives a theoretical performance evaluation of the system. The Cramer-Rao Lower Bound of the parameter estimates of a linear segment is derived. In addition, the distortion effect caused by the time-delay bias, including rescaling, rotation and bending of the original object, is clarified. A simple method using a line segment is proposed to calibrate the bias term arised in the different parts of the system. Finally a simulation gives a numerical demonstration of the previous analysis and performance improvement of the suggested method.

  • 69. Gustafsson, Harald
    Human Voice Communications2001Report (Other academic)
    Abstract [en]

    Humans communicate by many means. The two senses most used for communication are the visual and auditory senses. These can be stimulated by other humans by using for example hand gestures, face expressions, speech and song. This report will give a short introduction to the auditory sense and to speech generation. The text is mainly an abridgement of the books "Hearing" edited by B. C. J. Moore, and "Acoustic Phonetics" by K. N. Stevens.

  • 70. Gustafsson, Harald
    Speech enhancement for mobile communications2000Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Three major applications of speech enhancement have been studied: noise reduction for mobile telephone handsfree accessories, noise reduction for handheld mobile telephones, and short delay noise reduction for frame-less telephone systems. During conversation, both hearing and speaking adapt to the background noise in a noisy environment. However, when the conversation takes place over the telephone disturbances are more annoying. The disturbances are a problem since the brain will not get the extra visual and other background information when interpreting the speech. The basis for all the methods presented in this thesis is a novel spectral subtraction algorithm, a speech enhancement method. The novel method improves the conventional spectral subtraction method by introducing causal properties and reducing fluctuations of the gain filter used in spectral subtraction. Further, one extra microphone is included to improve the estimate of the background noise. Frame-less communication systems are also considered . A new spectral subtraction method, which performs the speech enhancement filtering in the time-domain.

  • 71. Gustafsson, Harald
    et al.
    Claesson, Ingvar
    Lindgren, Ulf
    Speech Bandwidth Extension2001Conference paper (Refereed)
  • 72. Gustafsson, Harald
    et al.
    Claesson, Ingvar
    Nordholm, Sven
    Lindgren, Ulf
    Dual-Microphone Spectral Subtraction2000Report (Other academic)
    Abstract [en]

    Mobile phones are constantly decreasing in size, thereby complicating the acoustical functionality. Signal processing methods can be used to partially mitigate this problem. In this paper we suggest a method which uses multiple spectral subtraction functions and two microphones, introducing only a short signal delay. The idea is to use spectral subtraction methods to extract the noise as well as the speech during a single time-frame. The environment background noise may not be stationary, thereby limiting the method to only employ short estimates of the background noise signal. Results are presented for experiments in various environments, showing a reduced noise level in the processed signal compared with the un-processed signal, and with preserved speech quality.

  • 73. Gustafsson, Harald
    et al.
    Nordholm, Sven
    Claesson, Ingvar
    Noise Suppression Employing Short Delay Processing1999Conference paper (Refereed)
    Abstract [en]

    Many telecommunication systems allow only a short delay in the terminal. This restriction stems from the fact that a delay will disturb the conversation. Therefore only a short delay is permissible in noise reduction processing. The well-known spectral subtraction method has been modified to introduce only a short delay and a variance reduced gain function. This modification is accomplished by viewing the spectral subtraction as an FFT filter design method which designs a time-varying filter for each block of data. The design will, however, only give the amplitude function, thus in order to obtain a causal filter a phase must be imposed. The spectral subtraction will not provide any phase information hence a minimum phase is imposed on the filter. The designed filter is transformed to the time-domain and the actual noise reduction filtering will be performed in the time-domain with only a short delay. The proposed method reduces the noise by 10 dB with a maximum processing delay of 7 samples.

  • 74. Gustafsson, Harald
    et al.
    Nordholm, Sven
    Claesson, Ingvar
    Spectral Subtraction using Correct Convolution1999Conference paper (Refereed)
  • 75. Gustafsson, Harald
    et al.
    Nordholm, Sven
    Claesson, Ingvar
    Spectral Subtraction using Dual Microphones1999Conference paper (Refereed)
  • 76. Gustafsson, Harald
    et al.
    Nordholm, Sven
    Claesson, Ingvar
    Spectral Subtraction Using Reduced Delay Convolution and Adaptive Averaging2001In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 9, no 8, p. 799-807Article in journal (Refereed)
    Abstract [en]

    In handsfree speech communication the signal to noise ratio is often poor, which makes it difficult to have a relaxed conversation. By using noise suppression, the conversation quality can be improved. This paper describes a noise suppression algorithm based on spectral subtraction. The method employs a noise and speech dependent gain function for each frequency component. Proper measures have been taken to obtain a corresponding causal filter and also to ensure that the circular convolution originating from FFT filtering yields a truly linear filtering. A novel method that uses spectrum-dependent adaptive averaging to decrease the variance of the gain function is also presented. The results show a 10-dB background noise reduction for all input SNR situations tested in the range -6 to 16 dB, as well as improvement in speech quality and reduction of noise artifacts as compared with conventional spectral subtraction methods.

  • 77. Gustafsson, Harald
    et al.
    Nordholm, Sven
    Claesson, Ingvar
    Spectral Subtraction with Adaptive Averaging of the Gain Function1999Conference paper (Refereed)
  • 78. Gustafsson, Ronnie
    Combating Intersymbol Interference and Cochannel Interference in Wireless Communication Systems2003Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Over the last decade the world has witnessed explosive growth in the use of wireless mobile communications. Looking around we find users with mobile phones, wireless PDAs, pagers, MP3 players, and wireless headphones to connect to these devices - a small testament of the impact of wireless communications on our daily lives. In addition the burst of new technologies such as Bluetooth and other short-range wireless communications are encouraging the further development of a wide variety of distributed wireless devices. Two major impediments to high-performance digital wireless communication systems are intersymbol interference (ISI) and cochannel interference (CCI). ISI is caused by the frequency selectivity (time dispersion) of the channel due to multipath propagation. Equalizers can be used to compensate for these channel distortions. One may design an equalizer given the received signal, or one may first estimate the channel impulse response and then design an equalizer based on the estimated channel. CCI, on the other hand, arises from cellular frequency reuse and thus limits the quality and capacity (number of users) of wireless networks. CCI can be reduced by the use of adaptive antenna arrays (also known as "smart antennas"). These systems utilize an array of antenna elements that provide directional (spatial) information about the received signals. Since the desired signal and unwanted cochannel interferers generally arrive from different directions, an adaptive beamforming algorithm can adjust the spatial gain to enhance the desired signal and mitigate the cochannel interferers. In this Thesis, we describe receiver architectures and adaptive signal processing algorithms designed to compensate for the ISI and CCI in wireless communication systems.

  • 79.
    Gustafsson, Ronnie
    et al.
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Kongolo, Didier
    Blekinge Institute of Technology, Department of Telecommunications and Signal Processing.
    Utvärdering av en beslutsåterkopplad kanalestimator för tredje generationens mobiltelefonisystem1999Independent thesis Advanced level (degree of Master (One Year))Student thesis
    Abstract [en]

    Examensarbetets syfte är att jämföra två metoder för estimering av framkanalen (basstation till mobiltelefon) i Wideband Code Division Multiple Access (WCDMA). Den första estimeringsmetoden, Optimal Feedforward Channel Estimation (OFCE), är en konventionell estimeringsmetod som baserar estimaten på för sändare och mottagare känd information, så kallade piloter. Den andra metoden, Decision Directed Channel Estimation (DDCE), är en vidareutveckling av OFCE. Den är besluts\-åter\-kopplad och använder även användardata vid estimeringen. Genom simuleringar kan man dra slutsatsen att DDCE, i jämförelse med OFCE, inte tillåter en minskning i signal-brus-förhållandet (SNR) då bitfelssannolikheten (BER) är stor. En förbättring går att mäta först vid mycket lägre BER än vad som är normalt i verkliga sammanhang. Slutsatsen blir att DDCE är komplex, väldigt beräkningskrävande, och ger liten förbättring på SNR vid normala BER, vilket gör det befogat att fråga sig om det är möjligt att använda metoden i ett mobiltelefonisystem.

  • 80. Gustafsson, Ronnie
    et al.
    Mohammed, Abbas
    A new pilot-signal based space-time adaptive algorithm2003Conference paper (Refereed)
  • 81. Gustafsson, Ronnie
    et al.
    Mohammed, Abbas
    Simulation of Wireless Fading Channels2003Report (Refereed)
    Abstract [en]

    Future services in wireless communications will increase the need for high bit rates in the system because of the use of ``wideband'' contents such as streaming video and audio. For example, there might be services where the users can download movies to the car theatre system, or where a user can react to the doorbell ringing even though he or she is miles away from home. It will be possible to run business software remotely using mobile devices, removing the need for ``dumb down'' software in the terminal devices. Also, users might want to be able to seamlessly roam between different air interfaces or standards using the same device. Some basic services are already implemented in the existing 2G systems such as GSM or IS-136, more services are planned for the new 3G systems and other advanced features have to wait until 2011 when 4G is scheduled for release. The demands for higher bit rates combined with the ever-increasing number of users, however, introduces the need for clever and efficient usage of the limited resource of the wireless channel. Two major impediments to high-performance wireless communication systems are intersymbol interference (ISI) and cochannel interference (CCI). ISI is caused by the frequency selectivity (time dispersion) of the channel due to multipath propagation and CCI is due to cellular frequency reuse. Equalizers can be used to compensate for ISI and CCI can be reduced by the use of adaptive antenna arrays (also known as "smart antennas"). The smart antenna utilizes an array of antenna elements that provide directional (spatial) information about the received signals. Since the desired signal and unwanted cochannel interferers generally arrive from different directions, an adaptive beamforming algorithm can adjust the spatial gain to enhance the desired signal and mitigate the cochannel interferers. In this Report we discuss the basic propagation mechanisms affecting the performance of wireless communication systems. We also present the implementation of a simulator which takes these mechanisms into account and verifies its performance for different channels. We also introduce basic equalization and beamforming concepts. Finally, we evaluate the recursive least squares (RLS) equalizer and receiver structures and assess their performance in combating the destructive effects of the channel.

  • 82. Gustafsson, Ronnie
    et al.
    Mohammed, Abbas
    Simulation of Wireless Fading Channels using Matlab2003Conference paper (Refereed)
    Abstract [en]

    Future services in wireless communications will increase the need for high bit rates in the system because of the use of "wideband" contents such as streaming video and audio. There might for example be services where the users can download movies to the car theatre system. The demands for higher bit rates combined with the ever-increasing number of users, however, introduces the need for clever and efficient usage of the limited resource of the wireless channel, by the use of properly designed transceivers. Simulations may be employed to investigate the performance of such transceivers in a cost effective way. In this paper, we discuss the basic propagation mechanisms affecting the performance of a wireless communication system and give hints and tips of how to implement these mechanisms in a simulator using Matlab.

  • 83. Gustafsson, Ronnie
    et al.
    Mohammed, Abbas
    Claesson, Ingvar
    A Combined Channel Estimation Algorithm for Coherent Detection in Mobile Communication Systems2002Conference paper (Refereed)
    Abstract [en]

    In this paper, we propose a coherent detection method that can reduce the estimation errors of the carrier phase resulting from Gaussian noise in communication systems where pilot symbol-assisted equalization is employed to compensate for fading distortion. The proposed approach is to use a combined algorithm consisting of the conventional feed-forward channel estimation and decision feedback channel estimation. Simulation results show that the proposed algorithm provides better performance than the conventional fading channel estimation approach.

  • 84. Gustafsson, Ronnie
    et al.
    Mohammed, Abbas
    Grbic, Nedelko
    Claesson, Ingvar
    A Block Based Eigenvector Equalization for Time: Varying Channels2002Conference paper (Refereed)
    Abstract [en]

    In this paper we investigate the EigenVector Algorithm for Blind Equalization (EVA), and extend its application to the equalization of time-varying fading channels. The modified iterative algorithm, termed Block-Based EVA (BBEVA), is evaluated in terms of ISI suppression, MSE and by examining the signal constellation at the output of the equalizer. Simulation results show that the BBEVA performs better than the non-blind Least Mean Squares (LMS) algorithm.

  • 85. Gustavsson, Ingvar
    A Remote Access Laboratory for Electrical Circuit Experiments2003In: International journal of engineering education, ISSN 0949-149X, Vol. 19, no 3, p. 409-419Article in journal (Refereed)
    Abstract [en]

    Many laboratory experiments in electrical engineering courses can be performed remotely using real equipment in a laboratory. Traditional circuit theory experiments have been conducted over the Internet at Blekinge Institute of Technology (BTH) in Sweden using the same experimental set-up from different locations simultaneously. The circuits are formed using remotely controlled switch matrices. The instruments and switch matrices used are computer-based PXI (PCI Extensions for instrumentation) devices which have virtual front panels that can be displayed on a remote PC. This approach is neither a simulation nor a SCADA (Supervisory Control and Data Acquisition) application. The students control the instruments in the same way as they would in a local laboratory. The only difference is that they do not form the circuits and connect the test probes manually. These laboratory experiments have been used successfully in undergraduate engineering education at BTH and at Luleå University of Technology, Sweden using a lab server at BTH. Two transducer laboratory exercises are also available for more experienced students, who mostly welcome the chance of doing the experiments from home at any convenient time. These exercises contain comparatively slow mechanical movements allowing only one user to be logged on and controlling the experiments at once. Video transmission is provided so other users can follow what is happening and also perform parts of the experiments.

  • 86. Gustavsson, Ingvar
    A Remote Lab for Electrical Experiments2003In: Lab on the Web -- Running Real Electronics Experiments via the Internet / [ed] SHUR, T. A. FJELDLY and M. S., New York : John Wiley & Sons , 2003Chapter in book (Other academic)
    Abstract [en]

    Traditional laboratory experiments in electrical engineering are provided remotely over the Internet using lab equipment at Blekinge Institute of Technology (BTH) in Sweden for distant learning students as well as for other people around the globe. The circuit theory experiments offered can be conducted from different locations simultaneously. Two transducer exercises are also available. These contain comparatively slow mechanical movements, and only one user at a time can be logged on and control the experiments. Video transmission is also provided to monitor the movements. Both types of laboratory exercises are used in undergraduate education at BTH. The address of the home page of the laboratory is http://www.its.bth.se/distancelab/english/. There you will find software packages to be used to access the lab equipment and also instructions etc.

  • 87. Gustavsson, Ingvar
    A Remote Laboratory for Electrical Experiments2002Conference paper (Refereed)
    Abstract [en]

    Many laboratory experiments in electrical engineering courses can be performed remotely using real equipment. Conventional electrical circuit experiments have been conducted over the Internet at BTH (Blekinge Tekniska Högskola: The Blekinge Institute of Technology) in Sweden from different locations simultaneously using an experimental hardware setup in a closed room at BTH. This is neither a simulation nor a SCADA (Supervisory Control and Data Acquisition) application. The students control the instruments in the same way as they would in the local laboratory. The only difference is that they do not form the circuits and connect the test probes manually. Not only the experiment itself is important but also the measurement procedure and the handling of the instruments.

  • 88. Gustavsson, Ingvar
    A traditional electronics laboratory with Internet access2003Conference paper (Refereed)
    Abstract [en]

    An electronics laboratory with Internet access emulating a traditional university laboratory for undergraduate education has been set up at Blekinge Institute of Technology (BTH) in Sweden. People located in different places around the globe can perform experiments simultaneously using client PCs connected to the BTH lab server via the Internet. Only a 56 kbit/s modem and MS Internet Explorer are required. The client software can be downloaded from the laboratory web site. Equipment such as power supply, function generator, digital multi-meter, oscilloscope, and breadboard are provided. A breadboard is used by the students to form the circuits and connect the test probes. This paper describes the laboratory and discusses some implementation issues.

  • 89. Gustavsson, Ingvar
    Remote Laboratory Experiments in Electrical Engineering Education2002Conference paper (Refereed)
    Abstract [en]

    A remote or online laboratory is a laboratory where you can access experiments and instruments or other equipment from outside over the Internet. Laboratories for undergraduate education or vocational training in basic electrical engineering are easy to control remotely. You cannot see the electrical current with your eyes or hear it so there is no need for sound or video transmission. Computer-based instruments do not have any control buttons or displays on the front panel. They have virtual front panels on the host computer only and those panels can be moved to a remote computer screen. However, the manual forming of circuits and connecting of test probes cannot be transferred. These actions have to be performed in another way in a remote laboratory. Remotely controllable switch matrices must be used. In the remote laboratory at BTH (Blekinge Institute of Technology) a client/server architecture is used. The student makes all the settings wanted on her client computer and then sends them to a lab server. The server makes measure-ments requested and returns the data obtained. The whole procedure takes only a second or two. A number of clients can access the experiments simultaneously. The laboratory is used in ordinary courses for on-campus students. They ac-cess the laboratory from a computer hall or from elsewhere outside the university. Due to the low number of bytes trans-ferred a 56 kbit modem will do.

  • 90. Gustavsson, Ingvar
    Traditional laboratory exercises and remote experiments in electrical engineering education2003Conference paper (Refereed)
    Abstract [en]

    Laboratory work is recognized as an efficient method for students to assimilate knowledge and to develop skills for solving real world problems. The Internet provides new opportunities for remote experimentation. Laboratory exercises in electrical engineering courses such as circuit theory and basic electronics can be performed remotely using real equipment. What equipment is required for remote experiments? Is it possible to design and implement a traditional laboratory with Internet access only? A traditional undergraduate electronics laboratory at Blekinge Institute of Technology (BTH), Sweden, makes eight identical lab stations available. Each station is equipped with a breadboard, some desk top instruments and a power supply. Experiments on electrical circuits have been conducted over the Internet using experimental hardware located in a small closed room at BTH. This tiny laboratory provides one lab station equipped with computer-based instruments and a remotely controlled switch matrix to replace the traditional breadboard. The matrix makes it possible to make all the necessary connections needed to form a circuit and to connect test probes in a fraction of a second. Is it possible to use time sharing to enable students to perform experiments simultaneously at eight different locations? How can I argue that a remote experiment is not a simulation? This paper addresses these and other similar questions and discusses some implementation issues. The address of the laboratory home page is http://www.its.bth.se/distancelab/english/.

  • 91. Gustavsson, Ingvar
    User-defined Electrical Experiments in a Remote Laboratory2003Conference paper (Refereed)
    Abstract [en]

    Laboratory exercises in electrical engineering courses can be performed remotely using real equipment. A number of user-defined experiments on electrical circuits have been conducted over the Internet at Blekinge Institute of Technology (BTH), Sweden; the experiments have been carried out in different locations simultaneously using the same experimental hardware located in a small closed laboratory at BTH. The laboratory provides a remotely controlled switch matrix, two function generators, a digital multi-meter, and an oscilloscope. The matrix replaces the traditional breadboard and students and other users around the globe use it to form circuits from components mounted in component holders in the matrix. It has five nodes; a jumper lead or up to four components can be connected between each pair of nodes. The laboratory supervisor or a teacher can easily swap components. Users control the instruments using virtual front panels in the same way as they had done earlier in the local laboratory; the only difference is that they no longer form the circuits and connect the test probes manually. Circuits are defined using PSpice compatible net lists. The sources and components available in the laboratory are listed in a library. This library can be added to the libraries in, for example, the evaluation version of PSpice. Students can, within certain limits, modify the circuits shown in the laboratory instruction manuals or even design circuits of their own. A virtual laboratory instructor checks the circuits formed automatically before the voltage is applied to avoid possible damage. Is it possible to establish a reasonable balance between the teachers’ needs and the complexity of the hardware? Can the virtual instructor check the circuits formed without making advanced calculations or simulations? This paper addresses these questions and discusses implementation issues.

  • 92. Gustavsson, Jan-Olof
    et al.
    Nordebo, Sven
    Börjesson, Per Ola
    Simultaneous Channel and Symbol Maximum Likelihood Estimation1999Conference paper (Refereed)
  • 93. Haan, Jan Mark de
    Filter Bank Design for Subband Adaptive Filtering2001Licentiate thesis, comprehensive summary (Other academic)
    Abstract [en]

    Adaptive filtering is an important subject in the field of signal processing and has numerous applications in fields such as speech processing and communications. Examples in speech processing include speech enhancement, echo- and interference- cancellation, and speech coding. Subband filter banks have been introduced in the area of adaptive filtering in order to improve the performance of time domain adaptive filters. The main improvements are faster convergence speed and the reduction of computational complexity due to shorter adaptive filters in the filter bank subbands. Subband filter banks, however, often introduce signal degradations. Some of these degradations are inherent in the structure and some are inflicted by filter bank parameters, such as analysis and synthesis filter coefficients. Filter banks need to be designed so that the application performance degradation is minimized. The presented design methods in this thesis aim to address two major filter bank properties, transmission delay in the subband decomposition and reconstruction as well as the total processing delay of the whole system, and distortion caused by decimation and interpolation operations. These distortions appear in the subband signals and in the reconstructed output signal. The thesis deals with different methods for filter bank design, evaluated on speech signal processing applications with filtering in subbands. Design methods are developed for uniform modulated filter banks used in adaptive filtering applications. The proposed methods are compared with conventional methods. The performances of different filter bank designs in different speech processing applications are compared. These applications are acoustic echo cancellation, speech enhancement including spectral estimation, subband beamforming, and subband system identification. Real speech signals are used in the simulations and results show that filter bank design is of major importance.

  • 94. Haan, Jan Mark de
    et al.
    Claesson, Ingvar
    Gustafsson, Harald
    Least Squares Design of Nonuniform Filter Banks with evaluation Seech Enhancement2003Conference paper (Refereed)
    Abstract [en]

    This paper presents a method for least squares design of nonuniform filter banks for application in subband signal processing. Design objectives aim to optimize the filter bank frequency response while minimizing subband and output aliasing. Aliasing is minimized although magnitude and phase changes affect the aliasing terms. Filter banks with increasing bandwidth are designed with the proposed method and evaluated in speech enhancement using a spectral subtraction algorithm. When using a nonuniform frequency resolution approximating that of the human auditory system it is shown that an increased noise reduction and SNR improvement is achieved while maintaining the speech quality for a fixed number of frequency-bands.

  • 95. Haan, Jan Mark de
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Filter Bank Design for Subband Adaptive Filtering Applications2001Report (Other academic)
    Abstract [en]

    Subband adaptive filtering has been proposed to improve the convergence rate while reducing the computational complexity associated with time domain adaptive filters. Subband processing introduces transmission delays caused by the filters in the filter bank and signal degradations due to aliasing effects. One way to reduce the aliasing effects is to impose oversampling in subbands rather than critical sampling and thus reduce signal degradation. By doing so, additional degrees of freedom are introduced for the design of filter banks which may be optimally exploited. In this report, a design method for modulated uniform filter banks with any oversampling is suggested, where the analysis filter bank delay and the total filter bank delay may be specified, and where the aliasing and magnitude/phase distortions are minimized.

  • 96. Haan, Jan Mark de
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Nordholm, Sven
    Design and Evaluation of Nonuniform DFT Filter Banks in Subband Microphone Arrays2002Conference paper (Refereed)
    Abstract [en]

    This paper presents a method for the design of nonuniform DFT filter banks for subband beamforming. Filter banks designed with the method are evaluated in subband beamforming in a real-world microphone array application. Different source positions in array applications give rise to different signal delays, which means that adaptive beamformers in the subbands alter the phase information of the subband signals in order to extract the source from a noisy background. Phase alterations in the subbands lead to signal degradations when perfect reconstruction filter banks are used for the subband decomposition and reconstruction. The objective of the proposed design is to minimize the magnitude of all aliasing components individually, such that aliasing distortion is minimized although phase alterations occur in the subbands. The proposed method is evaluated in a car hands-free mobile telephony environment with real speech signals and the results show that the performance can be increased by several decibel when using nonuniform filter banks instead of uniform filterbanks while maintaining the length of the subband filters.

  • 97. Haan, Jan Mark de
    et al.
    Grbic, Nedelko
    Claesson, Ingvar
    Nordholm, Sven
    Filter Bank Design for Subband Adaptive Microphone Arrays2003In: IEEE transactions on speech and audio processing, ISSN 1063-6676, E-ISSN 1558-2353, Vol. 11, no 1, p. 14-23Article in journal (Refereed)
    Abstract [en]

    This paper presents a new method for the design of oversampled uniform DFT-filter banks for the special application of subband adaptive beamforming with microphone arrays. Since array applications rely on the fact that different source positions give rise to different signal delays, a beamformer alters the phase information of the signals. This in turn leads to signal degradations when perfect reconstruction filter banks are used for the subband decomposition and reconstruction. The objective of the filter bank design is to minimize the magnitude of all aliasing components individually, such that aliasing distortion is minimized although phase alterations occur in the subbands. The proposed method is evaluated in a car hands-free mobile telephony environment and the results show that the proposed method offers better performance regarding suppression levels of disturbing signals and much less distortion to the source speech.

  • 98. Haan, Jan Mark de
    et al.
    Grbic, Nedelko
    Nordholm, Sven
    Claesson, Ingvar
    Design of oversampled uniform dft filter banks with delay specification using quadratic optimization2001Conference paper (Refereed)
    Abstract [en]

    Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. Subband processing introduces transmission delays caused by the filter bank and signal degradations due to aliasing effects. One efficient way to reduce the aliasing effects is to allow a higher sample rate than critically needed in the subbands and thus reduce subband signal degradation. We suggest a design method, for uniform DFT filter banks with any oversampling factor, where the total filter bank group delay may be specified, and where the aliasing and magnitude/phase distortions are minimized.

  • 99. Haan, Jan Mark de
    et al.
    Gustafsson, Harald
    Spectral Subtraction Using Model-based Spectra1999Conference paper (Refereed)
    Abstract [en]

    When using a mobile phone with a handsfree equipment in a vehicle it is difficult to have a relaxed conversation due to the high noise level. The noisy environment gives raise to a poor signal to noise ratio since the handsfree microphone picks up the background noise as well as the speech. By using speech enhancement processing, the quality of the conversation will be enhanced. This paper introduces a speech enhancement algorithm based on the spectral subtraction method. The method uses estimates of the noise spectrum and the noisy speech spectrum to form an SNR-based gain function, which is used for filtering the microphone signal. There are two different classes of spectrum estimation techniques non-parametric and parametric. Parametric estimation methods are based on a model of the underlying signal in order to estimate the spectrum.Furthermore a tool for assessing the accuracy of the estimations in situations when few data is available is applied, known as the bootstrapmethod. The spectral subtraction method, combined with different spectrum estimation techniques, are evaluated using real-world signals recorded in a car.

  • 100. Haan, Jan Mark de
    et al.
    Larson, Lars-Olof
    Claesson, Ingvar
    Filter Bank Design for Delayless Subband Adaptive Filtering Structures with Subband Weight Transformation2002Conference paper (Refereed)
    Abstract [en]

    Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. Subband processing introduces transmission delays and signal degradations due to aliasing effects. In order to overcome the delays, delayless adaptive filtering has been introduced where the adaptive algorithm is working in subbands while the signal filtering is performed in fullband. A design method for filter banks with any oversampling factor is utilized, where the total filter bank group delay may be specified, and where the aliasing and magnitude/phase distortions are minimized. This paper investigates the performance of the filter banks in delayless adaptive filtering applied in an acoustic echo cancellation scenario for telephony conferencing

123456 51 - 100 of 279
CiteExportLink to result list
Permanent link
Cite
Citation style
  • apa
  • harvard1
  • ieee
  • modern-language-association-8th-edition
  • vancouver
  • Other style
More styles
Language
  • de-DE
  • en-GB
  • en-US
  • fi-FI
  • nn-NO
  • nn-NB
  • sv-SE
  • Other locale
More languages
Output format
  • html
  • text
  • asciidoc
  • rtf