FFmpeg
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Go to the source code of this file.
Functions | |
id | N () |
FFT size for | synthesis (even) H |
N, 1 | zeros () |
overlap indexes | sw (ovidx) |
Variables | |
function [y] | |
positive part of the spectrum | Ns = 1024 |
analysis synthesishop size | soundlength = length(x) |
initialize buffer for FFT Create a loop to step through the sound array x initializing the loop | hNs = Ns/2 |
half synthesis window size | hM = (M-1)/2 |
half analysis window size used to overlap windows | pin = max(hNs+1,1+hM) |
initialize sound pointer to middle of analysis window | pend = soundlength-max(H,hM) |
last sample to start a frame | y = zeros(soundlength,1) |
initialize output array | w = w/sum(w) |
normalize analysis window | sw = zeros(Ns,1) |
ow = triang(2*H-1) | |
overlapping window | ovidx = Ns/2+1-hNs+1:Ns/2+H |
bh = blackmanharris(Ns) | |
Function Documentation
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virtual |
overlap indexes sw | ( | ovidx | ) |
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virtual |
Referenced by dct_unquantize_h263_axp(), encode_codeword(), and yae_load_frag().
Variable Documentation
Definition at line 25 of file harmonicmodel1.m.
function[y] |
Initial value:
%initializing values
id N()
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 it should directly call filter_frame on the corresponding output For a if there are queued frames already one of these frames should be pushed If the filter should request a frame on one of its repeatedly until at least one frame has been pushed Return values
Definition: filter_design.txt:216
Definition at line 1 of file harmonicmodel1.m.
initialize buffer for FFT Create a loop to step through the sound array x initializing the loop hNs = Ns/2 |
Definition at line 13 of file harmonicmodel1.m.
positive part of the spectrum Ns = 1024 |
Definition at line 5 of file harmonicmodel1.m.
ow = triang(2*H-1) |
Definition at line 22 of file harmonicmodel1.m.
Definition at line 17 of file harmonicmodel1.m.
Definition at line 16 of file harmonicmodel1.m.
Definition at line 7 of file harmonicmodel1.m.
Definition at line 21 of file harmonicmodel1.m.
initialize output array w = w/sum(w) |
Definition at line 20 of file harmonicmodel1.m.
Definition at line 19 of file harmonicmodel1.m.
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