[FFmpeg-devel] [PATCH] Lowpass functionality for lavc
Michael Niedermayer
michaelni
Thu Aug 14 16:40:27 CEST 2008
On Thu, Aug 14, 2008 at 04:29:15PM +0300, Kostya wrote:
> On Tue, Aug 12, 2008 at 07:33:43PM +0200, Michael Niedermayer wrote:
> > On Tue, Aug 12, 2008 at 08:11:49PM +0300, Kostya wrote:
> > > On Tue, Aug 12, 2008 at 07:58:19PM +0300, Kostya wrote:
> > > [...]
> > >
> > > oops, attached an old version of files, here're new ones
> >
> > [...]
> > > /** 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{
> > > int pos;
> > > float x[LOWPASS_FILTER_ORDER];
> > > float y[LOWPASS_FILTER_ORDER];
> > > }LPFilterState;
> >
> > I think these things could be moved to the c file
> >
> >
> > >
> > > /**
> > > * Initialize filter coefficients.
> > > *
> > > * @param order filter order
> > > * @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(int order, 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;
> > > float res;
> > >
> > > in *= coeffs->gain;
> > > //FIXME: made only for 4th order filter
> > > res = (s->x[s->pos&3] + in)*1 + (s->x[(s->pos+1)&3] + s->x[(s->pos+3)&3])*4 + s->x[(s->pos+2)&3]*6;
> > > for(i = 0; i < 4; i++)
> > > res += coeffs->c[i] * s->y[(s->pos + i)&3];
> > > s->x[s->pos&3] = in;
> > > s->y[s->pos&3] = res;
> > > s->pos++;
> > > return res;
> > > }
> >
> > i suspect this would be even slower than the moving of samples.
> > does gcc optimize the &3 stuff away? if not you will have to unroll it
> > by hand like i suggested, if OTOH gcc does optimize it away then its fine.
> > and with unroll i mean a function that works with 4 samples at a time.
> >
> >
> > [...]
> > > int ff_lowpass_filter_init_coeffs(int order, 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];
> >
> > memcpy
>
> Here's a new version that will require changing only implementation,
> interface will remain the same.
[...]
> /*
> * 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"
>
ok
> /**
> * Initialize filter coefficients.
> *
> * @param coeffs filter coefficients
> * @param freq input frequency (sample rate/2)
> * @param cutoff cutoff frequency
> *
> * @return pointer to filter coefficients structure or NULL if filter cannot be created
> */
> void* ff_lowpass_filter_init_coeffs(int order, int freq, int cutoff);
it should have some type not void* (and yes it can be done without putting the
whole struct in the header see libavutil/*.h)
you also forget to docuent order not that this is imporatnt i just noticed
>
> /**
> * Create new filter state.
> *
> * @param order filter order
> *
> * @return pointer to new filter state or NULL if state creation fails
> */
> void* ff_lowpass_filter_init_state(int order);
state and filter coefficients structure? why 2?
especially with void this is a recipe for confusion
[...]
> /*
> * 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"
>
> /**********************
> * TODO:
> * support filters with order != 4
> * calculate coefficients for filter instead of taking approximate ones from the table
> *********************/
>
> /** filter order */
> #define LOWPASS_FILTER_ORDER 4
>
> /**
> * 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 },
> };
>
ok
[...]
> void ff_lowpass_filter_free_coeffs(void *coeffs)
> {
> if(coeffs)
> av_free(coeffs);
> }
>
> void ff_lowpass_filter_free_state(void *state)
> {
> if(state)
> av_free(state);
> }
the if are useless
>
> void ff_lowpass_filter(void *coeffs, void *state, int size, int16_t *src, int sstep, int16_t *dst, int dstep)
> {
> int i, j;
> float in, res;
> LPFilterCoeffs *c = (LPFilterCoeffs*)coeffs;
> LPFilterState *s = (LPFilterState*) state;
>
> for(i = 0; i < size; i++, src += sstep, dst += dstep){
> for(j = 0; j < LOWPASS_FILTER_ORDER - 1; j++){
> s->x[j] = s->x[j+1];
> s->y[j] = s->y[j+1];
> }
luckily iam in good mood
please unroll the loop and remove this
its supposed to look something like
#define FILTER(a,b,c,d, src)
s->x[a]= *src++ * c->gain;
s->y[a]= s->x[a] + 123*s->y[a] + 321*s->y[b] + ...
*dst++= av_clip_int16(s->y[a]);
FILTER(0,1,2,3)
FILTER(1,2,3,0)
FILTER(2,3,0,1)
FILTER(3,0,1,2)
[...]
--
Michael GnuPG fingerprint: 9FF2128B147EF6730BADF133611EC787040B0FAB
The misfortune of the wise is better than the prosperity of the fool.
-- Epicurus
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