What exactly are you trying to do? You mentioned a C sharp filter? You actually want a filter that will reduce or eliminate just one note, leaving C and D intact?
You've got a little caught up in the clever stuff but have missed quite a bit out.
An audio waveform is an AC voltage, that swings above and below 0 Volts - so it goes positive and negative. With music, this is happening so fast that you can't really imagine it as a sine wave, for example, more like a few thousand or more sine waves all doing different things at the same time. Audio has two components Volume, as in how far away from 0 it goes. In a speaker circuit, it might easily be 70 or 80V peak to peak (as in highest positive going and highest negative going) It could also me a voltage measure in thousandths of a Volt - as in what might come out of a microphone. The other component is pitch - as in the frequency - so somewhere typically between 20 and 20,000Hz.
As said above, to make a filter - a device that reduces voltage at a specific frequency - you need capacitance and inductance. Filters have a 'quality' measured by how much attenuation they can manage, measured in dBs. If you plotted the attenuation against frequency, you could make a filter that removed 10dB at say 5000Hz. At 4500 and 5500Hz, it might only remove 7dB and at 4000 and 6000Hz maybe only 4 dB. So on a piano keyboard and you play C sharp, it would be very quiet. Play the C sharp an octave below and its normal volume. You might find that until you get to say, A, you can't hear a drop on volume. Bb would be quieter, B more quiet, same with C and then C sharp is quietest, then it starts to get louder again. If you increase the 'quality' of the filter, it gets narrower - so a filter has a frequency, an attenuation (or boost) amount and a Q adjustment - the narrowness (quality) of the filter. Simple filters can be very simple - 3 or 4 components, bit the Q of a circuit this simple will mean it's quite wide. Better filters mean they're more complex and clever.