echo cancellation using adaptive filters on tms320c6713
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pls send the required emospark paper on my mail id meghali.chakrabarty[at]gmail.com by tomo
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#2
Echo cancellation  methods in telephony to improve voice quality by preventing the echo from being created or deleted after it is already present. In addition to improving subjective quality, this process increases the capacity achieved by suppressing silence, preventing the echo travel through a network. These methods are commonly referred to as acoustic echo suppression (AES) and acoustic echo cancellation (AEC), and more rarely line echo cancellation (LEC). In some cases, these terms are more accurate, as there are several types and causes of echo with unique characteristics, including acoustic echo (sounds of a speaker reflected and recorded by a microphone, which can vary substantially over time) and echo Line Electrical impulses caused, for example, by the coupling between the sending and receiving cables, impedance impedances, electrical reflections, etc., which varies much less than acoustic echo). In practice, however, the same techniques are used to treat all types of echo, so an acoustic echo canceller can cancel line echo as well as acoustic echo. "AEC" in particular is commonly used to refer to echo cancellers in general, regardless of whether they were intended for acoustic echo, line echo or both.


Adaptive filters are playing a vital role in signal processing and archived engineering communication in order to filter unwanted signal, signal denaturation, signal enhancement, and so on. The main characteristic of the adaptive filter is the dynamic adjustment of the coefficients of the filter with respect to the one that helps a lot in signal processing applications. Adaptive filtering algorithms such as the LMS algorithm and standardized LMS algorithms (NLMS) are implemented with the floating-point DSP processor TMS320C6713 that uses the LabVIEW environment in real time. To test the functionality of the algorithms, the sinusoidal signal is added with noise and applied as input to the filter and the resulting output of decoupling is obtained with both algorithms. We implemented it with floating point digital signal processor TMS320C6713 using the LabVIEW environment in real time. Our objective is to reduce or filter noise using these algorithms and obtain performance metrics such as maximum output, mean square error (MSE), peak signal-to-noise ratio (PSNR) as part of the simulation results. The PSNR produced by the NLMS algorithm is obtained as 18,414 is very high compared to 3.28416 produced by the LMS algorithm. The TMS320C6713 DSP board is interconnected with the LabVIEW application using the Code Composer Studio software tool.
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