I have created a wav file (sampling rate of 44.1kHz at 32-bit floating point precision) consisting of 44.1k samples at 0.0 followed by 44.1k at 1.0. After which the file is played out from line-out using Media Player Classic and looped back into line-in. The resulting waveform is captured and inspected using Audacity.
The output waveform goes from 0.0 to 1.0, holds it there for a number of samples and some how "decays" back to 0.0. What is going on? And how do I recreate the desired step waveform where it will persist at 1.0 for the second half of the waveform?
The brand of the sound card used in question is Realtek, PC is an Acer TravelMate 4720.
PS: This may seems like a PC question but I suspect it has more to do with sound electronics used on sound cards.