[FFmpeg-devel] [PATCH] Lowpass functionality for lavc
Kostya
kostya.shishkov
Tue Aug 12 18:58:20 CEST 2008
On Tue, Aug 12, 2008 at 02:31:35PM +0200, Michael Niedermayer wrote:
> On Tue, Aug 12, 2008 at 02:23:55PM +0300, Kostya wrote:
> > $subj, made as a separate module.
>
> [...]
>
> > /**
> > * Initialize filter coefficients.
> > *
> > * @param coeffs filter coefficients
> > * @param freq input frequency (sample rate/2)
> > * @param cutoff cutoff frequency
> > *
> > * @return zero if filter creation succeeded, a negative value if filter could not be created
> > */
> > int ff_lowpass_filter_init_coeffs(LPFilterCoeffs *coeffs, int freq, int cutoff);
>
> the filter order should also be a parameter
added
> >
> > /**
> > * Filter input value.
> > *
> > * @param coeffs filter coefficients
> > * @param s filter state
> > * @param in input value
> > *
> > * @return filtered value
> > */
> > static av_always_inline float ff_lowpass_filter(LPFilterCoeffs *coeffs, LPFilterState *s, float in)
> > {
> > int i;
> > for(i = 0; i < LOWPASS_FILTER_ORDER; i++){
> > s->x[i] = s->x[i+1];
> > s->y[i] = s->y[i+1];
> > }
> > s->x[LOWPASS_FILTER_ORDER] = in * coeffs->gain;
> > //FIXME: made only for 4th order filter
> > s->y[LOWPASS_FILTER_ORDER] = (s->x[0] + s->x[4])*1
> > + (s->x[1] + s->x[3])*4
> > + s->x[2] *6
> > + coeffs->c[0]*s->y[0] + coeffs->c[1]*s->y[1]
> > + coeffs->c[2]*s->y[2] + coeffs->c[3]*s->y[3];
> > return s->y[LOWPASS_FILTER_ORDER];
> > }
>
> as already said in my other reply this should be unrolled so it needs no
> moving of samples.
unrolled
> > int ff_lowpass_filter_init_coeffs(LPFilterCoeffs *coeffs, int freq, int cutoff)
> > {
> > int i, j, size;
> > float cutoff_ratio;
> >
>
> > //since I'm too lazy to calculate coefficients, I take more or less matching ones from the table
> > //TODO: generic version
>
> hmm
Calculating actual filter parameters requires complex math
(I've looked into code of mkfilt by T.Fisher which I used for obtaining
filter coefficients and I should remember more of complex numbers than
Euler's formula to understand it).
I will do generic version when I get free time and refresh my algebra.
> --
> Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
>
> Asymptotically faster algorithms should always be preferred if you have
> asymptotical amounts of data
-------------- next part --------------
/*
* Lowpass IIR filter
* Copyright (c) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file lowpass.h
* lowpass filter interface
*/
#ifndef FFMPEG_LOWPASS_H
#define FFMPEG_LOWPASS_H
#include "avcodec.h"
/** filter order */
#define LOWPASS_FILTER_ORDER 4
/**
* IIR filter global parameters
*/
typedef struct LPFilterCoeffs{
float gain;
float c[LOWPASS_FILTER_ORDER];
}LPFilterCoeffs;
/**
* IIR filter state
*/
typedef struct LPFilterState{
float x[LOWPASS_FILTER_ORDER + 1];
float y[LOWPASS_FILTER_ORDER + 1];
}LPFilterState;
/**
* Initialize filter coefficients.
*
* @param coeffs filter coefficients
* @param freq input frequency (sample rate/2)
* @param cutoff cutoff frequency
*
* @return zero if filter creation succeeded, a negative value if filter could not be created
*/
int ff_lowpass_filter_init_coeffs(LPFilterCoeffs *coeffs, int freq, int cutoff);
/**
* Filter input value.
*
* @param coeffs filter coefficients
* @param s filter state
* @param in input value
*
* @return filtered value
*/
static av_always_inline float ff_lowpass_filter(LPFilterCoeffs *coeffs, LPFilterState *s, float in)
{
int i;
for(i = 0; i < LOWPASS_FILTER_ORDER; i++){
s->x[i] = s->x[i+1];
s->y[i] = s->y[i+1];
}
s->x[LOWPASS_FILTER_ORDER] = in * coeffs->gain;
//FIXME: made only for 4th order filter
s->y[LOWPASS_FILTER_ORDER] = (s->x[0] + s->x[4])*1
+ (s->x[1] + s->x[3])*4
+ s->x[2] *6
+ coeffs->c[0]*s->y[0] + coeffs->c[1]*s->y[1]
+ coeffs->c[2]*s->y[2] + coeffs->c[3]*s->y[3];
return s->y[LOWPASS_FILTER_ORDER];
}
#endif /* FFMPEG_LOWPASS_H */
-------------- next part --------------
/*
* Lowpass IIR filter
* Copyright (c) 2008 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file lowpass.c
* lowpass filter implementation
*/
#include "lowpass.h"
/**
* filter data for 4th order IIR lowpass Butterworth filter
*
* data format:
* normalized cutoff frequency | inverse filter gain | coefficients
*/
static const float lp_filter_data[][LOWPASS_FILTER_ORDER+2] = {
{ 0.5000000000, 9.398085e-01, -0.0176648009, 0.0000000000, -0.4860288221, 0.0000000000 },
{ 0.4535147392, 6.816645e-01, -0.4646665999, -2.2127207402, -3.9912017501, -3.2380429984 },
{ 0.4166666667, 4.998150e-01, -0.2498216698, -1.3392807613, -2.7693097862, -2.6386277439 },
{ 0.3628117914, 3.103469e-01, -0.0965076902, -0.5977763360, -1.4972580903, -1.7740085241 },
{ 0.3333333333, 2.346995e-01, -0.0557639007, -0.3623690447, -1.0304538354, -1.3066051440 },
{ 0.2916666667, 1.528432e-01, -0.0261686639, -0.1473794606, -0.6204721225, -0.6514716536 },
{ 0.2267573696, 6.917529e-02, -0.0202414073, 0.0780167640, -0.5277442247, 0.3631641670 },
{ 0.2187500000, 6.178391e-02, -0.0223681543, 0.1069446609, -0.5615167033, 0.4883976841 },
{ 0.2083333333, 5.298685e-02, -0.0261686639, 0.1473794606, -0.6204721225, 0.6514716536 },
{ 0.1587301587, 2.229030e-02, -0.0647354087, 0.4172275190, -1.1412129810, 1.4320761385 },
{ 0.1458333333, 1.693903e-02, -0.0823177861, 0.5192354923, -1.3444768251, 1.6365345642 },
{ 0.1133786848, 7.374053e-03, -0.1481421788, 0.8650973862, -1.9894244796, 2.1544844308 },
{ 0.1041666667, 5.541768e-03, -0.1742301048, 0.9921936565, -2.2090801108, 2.3024482658 },
};
int ff_lowpass_filter_init_coeffs(LPFilterCoeffs *coeffs, int freq, int cutoff)
{
int i, j, size;
float cutoff_ratio;
//since I'm too lazy to calculate coefficients, I take more or less matching ones from the table
//TODO: generic version
size = sizeof(lp_filter_data) / sizeof(lp_filter_data[0]);
cutoff_ratio = (float)cutoff / freq;
if(cutoff_ratio > lp_filter_data[0][0])
return -1;
for(i = 0; i < size; i++){
if(cutoff_ratio >= lp_filter_data[i][0])
break;
}
if(i == size)
i = size - 1;
coeffs->gain = lp_filter_data[i][1];
for(j = 0; j < 4; j++)
coeffs->c[j] = lp_filter_data[i][j+2];
return 0;
}
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