The objective of the research is to achieve a systematic combination of acoustic echo reduction components that together achieve a robust performance of the MCAEC system as a whole. Conventional approaches to the acoustic echo reduction system typically assume that individual components would perform ideally. For example, the adaptive algorithm for AEC is often developed in the absence of strong near-end signal, the algorithm for RES is often an added module that is developed as a separate noise reduction component, and the decorrelation procedure for MCAEC is yet another add-on module that simply introduces some form of distortion to the reference signal. The main challenge is in designing a consistent criterion across all modules that can be jointly optimized to form a more consistent framework for acoustic echo reduction. The decorrelation procedure can potentially benefit from the system approach as well if it is designed by taking the near-end listener into account. The MCAEC system should be optimized not only for the echo cancellation and suppression performance, but also for the reference signal quality after the added distortion from the decorrelation procedure. Finally, a tuning strategy is presented to jointly optimize the parameters across all modules using object criteria.

Ever since the birth of the telephony system, the problem with echoes, arising from impedance mismatch in 2/4-wire hybrids, or acoustic echoes where a loudspeaker signal is picked up by a closely located microphone, has been ever present. The removal of these echoes is crucial in order to achieve an acceptable audio quality for conversation. Today, the perhaps most common way for echo removal is through cancellation, where an adaptive filter is used to produce an estimated replica of the echo which is then subtracted from the echo-infested signal. Echo cancellation in practice requires extensive control of the filter adaptation process in order to obtain as rapid convergence as possible while also achieving robustness towards disturbances. Moreover, despite the rapid advancement in the computational capabilities of modern digital signal processors there is a constant demand for low-complexity solutions that can be implemented using low power and low cost hardware. This thesis presents low-complexity solutions for echo cancellation related to both the actual filter adaptation process itself as well as for controlling the adaptation process in order to obtain a robust system. Extensive simulations and evaluations using real world recorded signals are used to demonstrate the performance of the proposed solutions.

<p>Acoustic echo cancelation (AEC) composes a fundamental property of speech processing to enable a pleasant telecommunication conversation. Without this property of the telephone the communicator would hear an annoying echo of his own voice along with the speech from the other communicator. This would make a conversation through any telecommunication device an unpleasant experience.</p><p>AEC has been subject of interest since 1950s in the telecom industry and very efficient solutions were devised to cancel linear echo. With the advent of low cost hands free communication devices the issue of non linear echo became prominent because these devices use cheap loudspeakers that produce artifacts in addition to the desired sound which will cause non linear echo that cannot be cancelled by linear echo cancellers.</p><p>In this thesis a Harmonic Distortion Residual Echo Cancelation algorithm has been chosen for further investigations (HDRES). HDRES has many of those features that are desirable for an algorithm which is dealing with nonlinear acoustic echo cancelation, such as low computational complexity and fast convergence. The algorithm was first implemented in Matlab where it was tested and modified. The final result of the modified algorithm was then implemented in C and integrated with a complete AEC system. Before the implementation a number of measurements were done to distinguish the nonlinearities that were cause by the mobile phone loudspeaker. The measurements were performed on three different mobile pones which were documented to have problems with nonlinear acoustic echo.</p><p>The result of this thesis has shown that it might be possible to use an adaptive filter, which has both low complexity and fast convergence, in an operating AEC system. However, the request for such a system to work would be that a doubletalk detector is implemented along with the adaptive algorithm. That way the doubletalk situation could be found and the adaptation of the algorithm could be stopped. Thus, the major part of the speech would be saved.</p>

Acoustic echo cancelation (AEC) composes a fundamental property of speech processing to enable a pleasant telecommunication conversation. Without this property of the telephone the communicator would hear an annoying echo of his own voice along with the speech from the other communicator. This would make a conversation through any telecommunication device an unpleasant experience. AEC has been subject of interest since 1950s in the telecom industry and very efficient solutions were devised to cancel linear echo. With the advent of low cost hands free communication devices the issue of non linear echo became prominent because these devices use cheap loudspeakers that produce artifacts in addition to the desired sound which will cause non linear echo that cannot be cancelled by linear echo cancellers. In this thesis a Harmonic Distortion Residual Echo Cancelation algorithm has been chosen for further investigations (HDRES). HDRES has many of those features that are desirable for an algorithm which is dealing with nonlinear acoustic echo cancelation, such as low computational complexity and fast convergence. The algorithm was first implemented in Matlab where it was tested and modified. The final result of the modified algorithm was then implemented in C and integrated with a complete AEC system. Before the implementation a number of measurements were done to distinguish the nonlinearities that were cause by the mobile phone loudspeaker. The measurements were performed on three different mobile pones which were documented to have problems with nonlinear acoustic echo. The result of this thesis has shown that it might be possible to use an adaptive filter, which has both low complexity and fast convergence, in an operating AEC system. However, the request for such a system to work would be that a doubletalk detector is implemented along with the adaptive algorithm. That way the doubletalk situation could be found and the adaptation of the algorithm could be stopped. Thus, the major part of the speech would be saved.

Μελετάται η χρήση προσαρμοστικών αλγορίθμων οι οποίοι εφαρμόζονται στο πεδίο των συχνοτήτων και σκοπός τους είναι να ακυρωθεί σε όσον το δυνατόν μεγαλύτερο βαθμό η επίδραση της ηχούς σε ένα περιβάλλον τηλεδιάσκεψης. Όσον αφορά την προσομοίωση του τηλεπικοινωνιακού περιβάλλοντος αυτή θα γίνει με την χρησιμοποίηση κρουστικών αποκρίσεων οι οποίες θα λαμβάνονται χρησιμοποιώντας την μέθοδο των ψηφιακών κυματοδηγών. / Frequency domain adaptive filters are evaluated for use in a teleconferencing environment. The convergence rate, steady state, ability to track changes of the Frequency domain block quasi-Newton algorithm is compared to the Frequency domain block LMS (FD-BLMS)and time domain normalized LMS (TD-NLMS). Finally an algorithm for acoustic simulation of small rooms is derived in order to produce acoustic echo simulation data for use in the evaluation of the algorithms.

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.

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.

<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>

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.