隨著網路技術的發展，目前網路電話(VoIP)已有逐漸取代傳統電話的趨勢。尤其能夠允許多人同時在線上進行會談是其最大的優勢之一。但在多人參與網路會談時，因為聲音在空間中傳遞或反射等因素，使得由喇叭發出的聲音再次被麥克風收回，造成回音的產生。會談中只要有一位使用者的裝置發生回音時，回音訊號就會在與會者之間擴散，使得所有使用者均會受到影響，進而嚴重影響網路通話的進行。此狀況在參與會談人數越多時，發生機率越高，且對通話品質影響越嚴重。 傳統電話在一對一通話時，通常使用遠端回音消除機制(Near End Echo Canceller)，由接收端在接收聲音後先暫存在記憶體中再播放，再將麥克風擷取的聲音與事先暫存的訊號反向後混合，以抵銷回音。網路會談的環境下，由於沒有標準的聽筒設備，使得回音發生的時間難以預估。且多人參與的網路會談中，由於收聽者所聽到的聲音可能混合多個使用者說話的聲音與回音，使得回音訊號難以偵測。另外，由於網路傳輸的特性，回音訊號到達的時間與順序都難以預估，這使得回音消除機制在多人網路回談中經常失效。 本研究提出藉由語音動態偵測(Voice Activity Detection-VAD)的方式分辨回音訊號，藉由本研究所提出的語音能量VAD判定機制，能夠有效區別正常語音與回音的差異，即可有效的消除回音，同時發揮靜音抑制（Slience Suppression）的效果，阻擋不含語音內容的封包，降低網路頻寬耗用。本研究以自行開發的VoIP軟體進行實地測試實驗，實驗中顯示，我們的方法能消除85%以上的回音。 / With the prosperous development of Internet technology, traditional phone service is being replaced gradually byVoice-over-IP (VoIP) technology. One of the critical problem that is yet to be improved is the echo problem. Due to the difference in working environment, conventional echo cancellation technology may not work well on VoIP system. The echo problem is becoming more critical as the number of participants in a talk session increases. As long as one user fails to depress echos, every other participant in the conference will be infected. The more participant, the higher probability of echo infection. We propose an energy based Voice Activity Detection (VAD) mechnism that effectively differentiate echo from speech signal. Our VAD algrouthm records a user’s speech volume, and based on this information to determine whether the frame is echo or not. By applying this mechnism to network conference, we can filter out echo frames and suppress slience at same time to save bandwidth consumption. We experimented on a self-developed VoIP software platform, the experiment result shows that our method can eliminate more than 85% of the echo.

Multicarrier (MC) Modulation attracts a lot of attention for high speed wireless transmissions because of its capability to cope with frequency selective fading channels turning the wideband transmission link into several narrowband subchannels whose equalization, in some situations, can be performed independently and in a simple manner. Nowadays, orthogonal frequency division multiplexing (OFDM) with the cyclic prefix (CP) insertion is the most widespread modulation among all MC modulations, and this thanks to its simplicity and its robustness against multipath fading using the cyclic prefix. Systems or standards such as ADSL or IEEE802.11a have already implemented the CP-OFDM modulation. Other standards like IEEE802.11n combine CP-OFDM and multiple-input multiple-output (MIMO) in order to increase the bit rate and to provide a better use of the channel spatial diversity. Nevertheless, CP-OFDM technique causes a loss of spectral efficiency due to the CP as it contains redundant information. Moreover, the rectangular prototype filter used in CP-OFDM has a poor frequency localization. This poor frequency localization makes it difficult for CP-OFDM systems to respect stringent specifications of spectrum masks.To overcome these drawbacks, filter-bank multicarrier (FBMC) was proposed as an alternative approach to CP-OFDM. Indeed, FBMC does not need any CP, and it furthermore offers the possibility to use different time-frequency well-localized prototype filters which allow much better control of the out-of-band emission. In the literature we find several FBMC systems based on different structures. In this thesis, we focus on the Saltzberg's scheme called OFDM/OQAM (or FBMC/OQAM). The orthogonality constraint for FBMC/OQAM is relaxed being limited only to the real field while for OFDM it has to be satisfied in the complex field. Consequently, one of the characteristics of FBMC/OQAM is that the demodulated transmitted symbols are accompanied by interference terms caused by the neighboring transmitted data in time-frequency domain. The presence of this interference is an issue for some MIMO schemes and until today their combination with FBMC remains an open problem.The aim of this thesis is to study the combination between FBMC and MIMO techniques, namely spatial multiplexing with ML detection. In the first part, we propose to analyze different intersymbol interference (ISI) cancellation techniques that we adapt to the FBMC/OQAM with MIMO context. We show that, in some cases, we can cope with the presence of the inherent FBMC interference and overcome the difficulties of performing ML detection in spatial multiplexing with FBMC/OQAM. After that, we propose a modification in the conventional FBMC/OQAM modulation by transmitting complex QAM symbols instead of OQAM ones. This proposal allows to reduce considerably the inherent interference but at the expense of the orthogonality condition. Indeed, in the proposed FBMC/QAM,the data symbol and the inherent interference term are both complex. Finally, we introduce a novel FBMC scheme and a transmission strategy in order to avoid the inherent interference terms. This proposed scheme (that we call FFT-FBMC) transforms the FBMC system into an equivalent system formulated as OFDM regardless of some residual interference. Thus, any OFDM transmission technique can be performed straightforwardly to the proposed FBMC scheme with a corresponding complexity growth. We develop the FFT-FBMC in the case of single-input single-output (SISO) configuration. Then, we extend its application to SM-MIMO configuration with ML detection and Alamouti coding scheme.

In the 1970’s, L. Lovász proved that two graphs G and H are isomorphic if and only if for every graph X , the number of homomorphisms from X → G equals the number of homomorphisms from X → H . He used this result to deduce cancellation properties of the direct product of graphs. We develop a result analogous to Lovász’s theorem, but in the class of graphs without loops and with weak homomorphisms. We apply it prove a general cancellation property for the strong product of graphs.

Há um grande interesse hoje em desenvolver terminais viva-voz que permitam aos participantes de uma conversa à distância contarem com um bom grau de naturalidade e inteligibilidade. O objetivo deste trabalho é apresentar solução para dois impedimentos que surgem quando se deseja projetar um terminal viva-voz para ser utilizado em automóveis: o eco acústico resultante do acoplamento entre microfone e alto-falante do terminal; e o ruído ambiente produzido por exemplo pelo vento, pneus e motor do veículo. A solução proposta envolve o uso de filtros adaptativos e alterações no espectro do sinal de voz para minimizar os problemas mencionados. Os aspectos teóricos são abordados de forma breve, sem deixar no entanto que nenhum detalhe importante fique de fora. Uma implementação prática e eficiente em processador digital de sinais é um dos destaques do trabalho. / There is currently great motivation in developing hands-free devices which offer users, engaged in a telephone conversation, a good level of naturalness and intelligibility. In this work, the goal is to present a solution for two well-known problems that occur when designing a hands-free device for use in automobile environments: (1) the acoustic echo coupling between microphone and speaker, and (2) the background noise generated for example by wind, tires and vehicle engine. The proposed solution includes adaptive filtering techniques and modifications in the speech signal spectrum, in order to minimize the two problems above. Theoretical issues are briefly analyzed, however the author believes no relevant detail is kept out. Highlighted in the report is a practical and efficient implementation of the algorithms in a modern digital signal processor.

Há um grande interesse hoje em desenvolver terminais viva-voz que permitam aos participantes de uma conversa à distância contarem com um bom grau de naturalidade e inteligibilidade. O objetivo deste trabalho é apresentar solução para dois impedimentos que surgem quando se deseja projetar um terminal viva-voz para ser utilizado em automóveis: o eco acústico resultante do acoplamento entre microfone e alto-falante do terminal; e o ruído ambiente produzido por exemplo pelo vento, pneus e motor do veículo. A solução proposta envolve o uso de filtros adaptativos e alterações no espectro do sinal de voz para minimizar os problemas mencionados. Os aspectos teóricos são abordados de forma breve, sem deixar no entanto que nenhum detalhe importante fique de fora. Uma implementação prática e eficiente em processador digital de sinais é um dos destaques do trabalho. / There is currently great motivation in developing hands-free devices which offer users, engaged in a telephone conversation, a good level of naturalness and intelligibility. In this work, the goal is to present a solution for two well-known problems that occur when designing a hands-free device for use in automobile environments: (1) the acoustic echo coupling between microphone and speaker, and (2) the background noise generated for example by wind, tires and vehicle engine. The proposed solution includes adaptive filtering techniques and modifications in the speech signal spectrum, in order to minimize the two problems above. Theoretical issues are briefly analyzed, however the author believes no relevant detail is kept out. Highlighted in the report is a practical and efficient implementation of the algorithms in a modern digital signal processor.

We consider discrete and continuous symmetric random variables X taking values in [0; 1], and thus having expected value 1/2. The main thrust of this investigation is to study the correlation between the variance, Var(X) of X and the value of the expected maximum E(Mn) = E(X1,...,Xn) of n independent and identically distributed random variables X1,X2,...,Xn, each distributed as X. Many special cases are studied, some leading to very interesting alternating sums, and some progress is made towards a general theory.

Evolution of wireless services enabled the development of the advanced applications and shifted the paradigms of research in this field from voice to data centric. Such services are spreading like wildfire between users and hence, increasing the demand for large bandwidth. However, the frequency spectrum that is suitable for wireless mobile communications is already assigned to particular services from 400 MHz to several GHz. Also, allocating a large chunk of band continuously from the same part of the spectrum may not be possible due to spectral crowd. Therefore, meeting the demand for high data rate requiring wireless services within the accessible spectrum range becomes a challenging problem. The spectrum allocation policies are discussed by regulatory authorities and academia, and the idea of spectrum sharing systems are addressed as a solution. For instance, heterogeneous networks (HetNets) increase the number of available resources and improve the spectrum accessing capabilities of the wireless communication systems. To achieve this, HetNet nodes are deployed within the coverage of the macrocell regions. Thus, spectral efficiency is boosted via spatial reuse of the same spectral resources. On the contrary, HetNets preclude to fully exploit the resources because of serious interference problems between macrocell and HetNet nodes. Thus, wireless networks of the future will observe interference from even a larger number of sources. Due to co-channel HetNet deployment and denser frequency reuse, interference cancellation is expected to have significant importance for future wireless communication systems. The occupied resources can also be reused as a solution by conducting advanced signal processing algorithms at the receiver to increase the spectral efficiency. While doing so, the proposed approaches are expected to be easily integrated with the existing complementary approaches to improve the capacity further. Besides, new deployment strategies that allow spectrum access for non-licensed users to achieve larger bandwidth become important to increase the spectral efficiency of the HetNets. Within the scope of the dissertation, new solutions are developed for the aforementioned problems of the next-generation wireless communication systems. First, an interference cancellation receiver that exploits the unique characteristics of current waveforms is developed in Chapter 2. Also the unknown model of interference is converted to a known model and new algorithms are proposed to recover the desired signal. Then, another perspective is brought into the subject by transforming the interference problem to an interference advantage in Chapter 3. The idea of co-existence of different types of signals are analyzed to bring another degree of freedom as a solution. The proposed approaches are integrated to the existing complementary approaches, such as interference coordination and power control, to improve the capacity further. Finally, a cooperation mechanism is suggested to facilitate the transmission of signal which has a large bandwidth by integrating the idle bands in Chapter 4. By this way, geo-spatially idle bands within the coverage area are utilized and spectral efficiency is increased.

<p>The acoustic echo problem arises whenever there is acoustic coupling between a loudspeaker and a microphone, such as in a teleconference system. This problem is traditionally solved by using an acoustic echo canceler (AEC), which models the echo path with adaptive filters. Long adaptive filters are necessary for satisfactory echo cancellation, which makes AEC highly computationally complex. Recently, a low-complexity echo suppression scheme was presented, the perceptual acoustic echo suppressor (PAES). Spectral modification is used to suppress the echoes, and the complexity is reduced by incorporating perceptual theories. However, under ideal conditions AEC performs better than PAES. </p><p>This thesis considers a hybrid system, which combines AEC and PAES. AEC is used to cancel low-frequency echo components, while PAES suppresses high-frequency echo components. The hybrid system is simulated and assessed, both through subjective listening tests and objective evaluations. The hybrid scheme is shown to have virtually the same perceived quality as a full-band AEC, while having a significantly lower complexity and a higher degree of robustness.</p>

On this thesis project, the LMS algorithm has been applied for speech noise filteringand different behaviors were tested under different circumstances by using Matlabsimulations and the LPC2378 ARM Processor, which does the task of filtering in realtime. The thesis project is divided into two parts: the theoretical and practical part. In the theoretical part there is a brief description of the different aspects of signalprocessing systems, filter theory, and a general description of the Least-Mean-SquareAdaptive Filter Algorithm. In the practical part of the report a general description of the procedure will besummarized, the results of the tests that were conducted will be analyzed, a generaldiscussion of the problems that were encounter during the simulations will be mention,and suggestion for the problems will be given.

In hands-free telephony and in teleconference systems, the main aim is to provide agood free voice quality when two or more people communicate from different places.The problem often arises during the conversation is the creation of acoustic echo. Thisproblem will cause the bad quality of voice signal and thus talkers could not hearclearly the content of the conversation, even thought lost the important information.This acoustic echo is actually the noise which is created by the reflection of soundwaves by the wall of the room and the other things exist in the room. The mainobjective for engineers is the cancellation of this acoustic echo and provides an echofree environment for speakers during conversation. For this purpose, scientists designdifferent adaptive filter algorithms. Our thesis is also to study and simulate theacoustics echo cancellation by using different adaptive algorithms.