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FFmpeg
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Go to the source code of this file.
Functions | |
| x | length () |
| overlap indexes | sw (ovidx) |
Variables | |
| function [y, yh, ys, fr0] | |
| M = length(w) | |
| analysis window size | Ns = 1024 |
| FFT size for synthesis | H = 256 |
| hop size for analysis and synthesis | N2 = N/2+1 |
| length of input sound array | hNs = Ns/2 |
| half synthesis window size | hM = (M-1)/2 |
| half analysis window size | pin = max(hNs+1,1+hM) |
| initialize sound pointer to middle of analysis window | pend = soundlength-max(hM,hNs) |
| last sample to start a frame | fftbuffer = zeros(N,1) |
| initialize buffer for FFT | yh = zeros(soundlength+Ns/2,1) |
| output sine component | ys = zeros(soundlength+Ns/2,1) |
| output residual component | w = w/sum(w) |
| normalize analysis window | sw = zeros(Ns,1) |
| ow = triang(2*H-1) | |
| overlapping window | ovidx = Ns/2+1-H+1:Ns/2+H |
| bh = blackmanharris(Ns) | |
| normalize synthesis window | wr = bh |
| sws = H*hanning(Ns) | |
| synthesis window for stochastic | i = 0 |
Function Documentation
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virtual |
| overlap indexes sw | ( | ovidx | ) |
Variable Documentation
Definition at line 31 of file hpsmodel.m.
Initial value:
if max(w)>1 w=0.9 *w/max(w)
About Git write you should know how to use GIT properly Luckily Git comes with excellent documentation git help man git shows you the available git< command > help man git< command > shows information about the subcommand< command > The most comprehensive manual is the website Git Reference visit they are quite exhaustive You do not need a special username or password All you need is to provide a ssh public key to the Git server admin What follows now is a basic introduction to Git and some FFmpeg specific guidelines Read it at least if you are granted commit privileges to the FFmpeg project you are expected to be familiar with these rules I if not You can get git from etc no matter how small Every one of them has been saved from looking like a fool by this many times It s very easy for stray debug output or cosmetic modifications to slip in
Definition: git-howto.txt:5
FFT size for synthesis(even) H
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several the filter must be ready for frames arriving randomly on any input any filter with several inputs will most likely require some kind of queuing mechanism It is perfectly acceptable to have a limited queue and to drop frames when the inputs are too unbalanced request_frame This method is called when a frame is wanted on an output For an input
Definition: filter_design.txt:216
phase spectrum(unwrapped) ploc
overlapping window(triangular window to avoid too much overlapping) ovidx
sound(x3, Fs)
these buffered frames must be flushed immediately if a new input produces new output(Example:frame rate-doubling filter:filter_frame must(1) flush the second copy of the previous frame, if it is still there,(2) push the first copy of the incoming frame,(3) keep the second copy for later.) If the input frame is not enough to produce output
Definition at line 1 of file hpsmodel.m.
Definition at line 16 of file hpsmodel.m.
Definition at line 37 of file hpsmodel.m.
Definition at line 14 of file hpsmodel.m.
Definition at line 15 of file hpsmodel.m.
| ow = triang(2*H-1) |
Definition at line 28 of file hpsmodel.m.
Definition at line 22 of file hpsmodel.m.
Definition at line 27 of file hpsmodel.m.
Definition at line 35 of file hpsmodel.m.
| output residual component w = w/sum(w) |
Definition at line 26 of file hpsmodel.m.
Definition at line 33 of file hpsmodel.m.
| initialize buffer for FFT yh = zeros(soundlength+Ns/2,1) |
Definition at line 24 of file hpsmodel.m.
| output sine component ys = zeros(soundlength+Ns/2,1) |
Definition at line 25 of file hpsmodel.m.
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