How can watashi be cool and hip and like minasan?

I maed this https://yuibox.yuisuki.net/ (NOT a shock image this time)

Its a free registration-based (no email needed) file uploader

feedback would be appreciated ヽ(´ー｀)ノ

Sincerely, ebook readers are the best thing ever. Virtually every book is digitized thanks to them and easily pirateable, meaning you have access to the world's full knowledge and universes of stories.

sitting in front of the pc no friends no nothing
i just want to cuddle with a girl

*drinks coffee*

cry my friends.

My phone broked

Cheer me up with anything

i want to rip my own face off

ｷﾀ━━━(ﾟ∀ﾟ)━━━!!

getting away with murder~!

longcat

ｷﾀ━━━(ﾟ∀ﾟ)━━━!!

I have implemented an improvement to the laryngograph onset detection procedure.

In typical laryngograph signals, like most signals, there is high-frequency noise. However, in addition to this high-frequency noise, there is also an abundance of low-frequency noise, usually much more than the high-frequency noise. This is due shifting of the neck and laryngograph. The standard DSP solution to this problem would be to use a highpass filter. However, the frequency of this noise, and thus the required transition band, are so low that it would render a standard single-stage FIR highpass filter impractical.

The complete procedure I have implemented is follows:
1. The signal is lowpass-filtered (to filter out the high-frequency noise).
2. Peaks (which correspond to glottal closure instants) are detected and interpolated using quadratic interpolation. The topographical prominence is also calculated for each peak, and peaks below a minimum prominence are discarded.
3. In between each adjacent pair of peaks, the minimum valley is found (also using quadratic interpolation).
4. An envelope is created using the natural cubic spline algorithm from this set of minimum valleys.
5. This envelope is subtracted from the original signal.
6. The signal is filtered again, replacing the old filtered signal.
7. Peaks are detected again, replacing the old set of peaks.
8. These peaks are grouped into regions using two heuristics: distance between subsequent peaks and change in subsequent distances.
9. Regions with only one peak are discarded.

Another approach would be to use two-stage downsampling. Then, it could be further lowpass-filtered to obtain fine-control (the new sample rate is chosen to be efficient, not necessarily precisely what is desired). These values could then be used in the same interpolation procedure; alternatively, they can be upsampled back to the original sampling rate (also using a two-stage process for efficiency reasons) and this signal subtracted from the laryngograph signal.

This approach is more correct in theory. On the other hand, this assumes a perfectly periodic signal. In reality, the laryngograph pulses vary. Furthermore, they are most definitely not period around the start and end of voiced regions. This aperiodicity will distort any filtering operation. On the other hand, the valleys in between pulse generally vary much less than the signal overall. Two stage downsampling is also less efficient than the spline approach.

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reply under so I can fucking ban you!

you better not coming to school tomorrow.