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chapter_01.tex
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@@ -6,12 +6,12 @@ Therefore, the improvement of implant performance in regard to the suppresion of
By addressing these challenges, this work aims to contribute to the next generation of cochlear implant technology, ultimately enhancing the auditory experience and quality of life for people with severe hearing impairments.
\subsection{Introduction to Cochlear Implant Systems}
A Cochlear Implant (CI) System is a specialized form of hearing aid, used to restore partly or complete deafness. In contrary to standard hearing aids, CI's do not just amplify the audio signal received by the ear, but stimulate the auditory nerve itself directly through electric pulses.\\ \\
Usually, a CI System consists out of an external processor (''audio processor'') receiving the ambient audio signal, processing it, and then transmitting it inductively via a transmission coil through the skin to the cochlear implant itself, implanted on the patient's skull (see figure \ref{fig:fig_snychrony}). The CI stimulates the auditory nerves inside the cochlear through charge pulses, thus enabling the patient to hear the received audio signal as sound.\\
Usually, a CI System consists out of an external processor (''audio processor'') receiving the ambient audio signal, processing it, and then transmitting it inductively via a transmission coil through the skin to the cochlear implant itself, implanted on the patient's skull (see figure \ref{fig:fig_synchrony}). The CI stimulates the auditory nerves inside the cochlear through charge pulses, thus enabling the patient to hear the received audio signal as sound.\\
\begin{figure}[H]
\centering
\includegraphics[width=0.6\linewidth]{Bilder/fig_synchrony.png}
\caption{Sketch of a MED-EL Synchrony Cochlear Implant with a Sonnet 3 Audio Processor \cite{source_synchrony}}
\label{fig:fig_snychrony}
\label{fig:fig_synchrony}
\end{figure}
The pulse transmission to the cochlear is realized through a silicone eletrode with embedded metal contacts. Said electrode is inserted into the cochlear through a drilled hole in the bone, where, depending on the insertion-depth, different contact-areas stimulate different parts of the frequency-spectrum of the hearing sense. The smaller end of the electrode-array inserted deep into the cochelar stimulate low-frequencies, whereas the larger beginning of the array stimulates high-frequencies. (see figure \ref{fig:fig_electrode}).
\begin{figure}[H]
@@ -25,10 +25,9 @@ As for any head worn hearing aid, the audio processor of a CI system does not on
A signal is a physical parameter (e.g. pressure, voltage) changing its value over time. The termn "Signal Interference" describes the overlapping of two or more signals resulting in a new signal. \\ \\A simple example of a desirable signal interference would be the sound generated by playing several strings of a guitar. Hitting one string results in a pure sine wave of a designated frequency (depending on which note is played), perceptible as sound. Hitting a chord (consisting of several strings), the seperate sine waves of the strings combine to a new signal through the process of signal interference - in this case a desired, harmonic sound. (see XXX)
In technical applications on the other hand, signal interference can cause a considerable degradation to the quality of the final signal, posing an additional challenge to aurally impaired people using a implant solution for rehabilitation. Fig XXX shows exemplary the comparison between the undisturbed audio signal of a person speaking and the audio signal of a person speaking disturbed by a strong static noise. Thus, the objetive of this thesis shall be the improvement of implant technology in regard of noise reduction.
\subsection{Implementation of Adaptive Noise Reduction in Cochlear Implant Systems}
The above problem statement of signal interference shows it#s significance in the improvement of CI systems. Whereas for persons with a healty hearing sense, the adding of a noise to a signal may just mean a decrease in hearing comfort, for aurally impaired people it can make the difference in understanding some important information or not.\\ \\
Adaptive Noise Reduction (ANR) (also commonly reffered as Adaptive Noise Cancellation (ANC)), is an advanced signal-processing technique that adjusts the parameters of a digital filter to surpress unwanted noise from a signal while preserving the desired target-filters. In contrary to static filters (like a low-pass filter), ANR uses realtime feedback to ajust said digital filter to adapt to the current cirumstances.\\ \\
The concept of ANR was already proposed in the the 1930´s by the german Physician Paul Lueg in form of two antiphase signales canelling eacht other. The fundamental Least-Mean-Square algroithm itself was developed towards the end of the 1950s. The first commercial use of ANR was seen in form of an aviation headset for pilots in 1989.\\ \\
The above problem statement of signal interference shows its significance in the improvement of CI systems. For persons with a healty hearing sense, the addition of noise to a observed signal may just mean a decrease in hearing comfort, wheras for aurally impaired people it can make the difference in the basic understanding of information. As everyday environments present fluctuating background noise - from static crowd chatter to sudden sounds of different characteristics — that can severely degrade speech perception, the ability to surpress noise is a crucial benefit for users of cochlear implant systems. \\ \\
Adaptive Noise Reduction (ANR) (also commonly reffered as Adaptive Noise Cancellation (ANC)), is an advanced signal-processing technique that adjusts the parameters of a digital filter to surpress unwanted noise from a signal while preserving the desired target-signal. In contrary to static filters (like a high- or low-pass filter), ANR uses realtime feedback to ajust said digital filter to adapt to the current cirumstances.\\ \\
The concept of ANR was already proposed in the the 1930´s by the german Physician Paul Lueg in form of two antiphase signales canelling eacht other, but the fundamental Least-Mean-Square algorithm to realize a technical implementation of ANR itself was developed not until towards the end of the 1950s. The first commercial use of ANR was finally seen in form of an aviation headset for pilots in 1989.\\ \\
The challenge in the implementation of ANR in CI systems lies in the limited capacities. As the CI system is powered by a small battery located in the audio processor, energy efficiency is crucial for a possible solution of the described problem of noise interference. Any approach to a reduction of interference noise must be highly optimized with regard to computing power and implemented on dedicated low-power hardware, being able to be powered within the limitations of a CI system.\\ \\
The main solution concept of this thesis is the optimization of the adaptive filter of the ANR algorithm in combination with the used low-power hardware. Its goal is, to deliver the best possible result in interference noise reduction while still being able to be powered by the limited resources of a CI system.
Different variants, like the fully adaptive filter, the hybrid static/adaptive filter and different optimization approaches of the latter one are low-level simulated on the dedicated DSP. Especially, the different optimization strategies of the hybrid static/adaptive filter algorithm shall be evaluated and compared in regard of their required computing power, and therefore, their required power consumption. Depending on the kind of interference noise, the frequency and the intensity, a promising optimization approach is the reduction of adaptation steps per sample while still maintaining an adequate quality of the filtered audio signal.\\ \\
The main solution concept of this thesis is the optimization of the adaptive filter of the ANR algorithm in combination with the used low-power hardware. Its goal is, to deliver the best possible result in interference noise reduction while still being able to be powered by the limited resources of a CI system. Different variants, like the fully adaptive filter, the hybrid static/adaptive filter and different optimization approaches of the latter one are low-level simulated on the dedicated DSP. Especially, the different optimization strategies of the hybrid static/adaptive filter algorithm shall be evaluated and compared in regard of their required computing power, and therefore, their required power consumption. Depending on the kind of interference noise, the frequency and the intensity, a promising optimization approach is the reduction of adaptation steps per sample while still maintaining an adequate quality of the filtered audio signal.\\ \\
Due to the fact, that the CI system is powered by a battery with a relatively small capacity, the firmware is required to work with the least power possible. Therefore, optimization in regard to a minimization of needed processor clocks is aimed for.

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chapter_02.tex
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\section{Theoretical Background}
\subsection{Fundamentals of signal theory and transfer functions, including a simple illus-
trative example}
\subsection{Introduction to ANR}
\subsection{Explanation of Finite Impulse Response (FIR) and Infinite Impulse Response
(IIR) filters}
\subsection{Introduction to the Least Mean Square (LMS) method for adaptive filtering}
\subsection{Problem analysis: Signal flow diagram showing the origin of the useful signal,
\subsection{Fundamentals of digital signal processing, transfer functions and filters}
\subsection{Introduction to Adaptive Nose Reduction}
\subsection{Explanation of Finite Impulse Response- and Infinite Impulse Response-filters}
\subsection{Introduction to the Least Mean Square algorithm for adaptive filtering}
\subsection{Signal flow diagram showing the origin of the useful signal,
noise signal, and their coupling}
\subsection{Derivation of the system’s transfer function based on the problem setup}
\subsection{Example applications and high-level simulations using Python}