817bf8483e
Fixes: CID1351398 Fixes: CID1351400 Reviewed-by: Paul B Mahol <onemda@gmail.com> Signed-off-by: Michael Niedermayer <michael@niedermayer.cc>
762 lines
23 KiB
C
762 lines
23 KiB
C
/*
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* Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others
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* Copyright (c) 2015 Paul B Mahol
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/intreadwrite.h"
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#include "libavutil/avstring.h"
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#include "libavutil/opt.h"
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#include "libavutil/parseutils.h"
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#include "avfilter.h"
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#include "internal.h"
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#include "audio.h"
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#define FILTER_ORDER 4
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enum FilterType {
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BUTTERWORTH,
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CHEBYSHEV1,
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CHEBYSHEV2,
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NB_TYPES
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};
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typedef struct FoSection {
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double a0, a1, a2, a3, a4;
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double b0, b1, b2, b3, b4;
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double num[4];
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double denum[4];
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} FoSection;
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typedef struct EqualizatorFilter {
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int ignore;
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int channel;
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int type;
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double freq;
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double gain;
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double width;
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FoSection section[2];
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} EqualizatorFilter;
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typedef struct AudioNEqualizerContext {
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const AVClass *class;
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char *args;
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char *colors;
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int draw_curves;
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int w, h;
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double mag;
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int fscale;
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int nb_filters;
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int nb_allocated;
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EqualizatorFilter *filters;
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AVFrame *video;
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} AudioNEqualizerContext;
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#define OFFSET(x) offsetof(AudioNEqualizerContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM
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#define V AV_OPT_FLAG_VIDEO_PARAM
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#define F AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption anequalizer_options[] = {
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{ "params", NULL, OFFSET(args), AV_OPT_TYPE_STRING, {.str=""}, 0, 0, A|F },
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{ "curves", "draw frequency response curves", OFFSET(draw_curves), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, V|F },
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{ "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, V|F },
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{ "mgain", "set max gain", OFFSET(mag), AV_OPT_TYPE_DOUBLE, {.dbl=60}, -900, 900, V|F },
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{ "fscale", "set frequency scale", OFFSET(fscale), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, V|F, "fscale" },
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{ "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, V|F, "fscale" },
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{ "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, V|F, "fscale" },
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{ "colors", "set channels curves colors", OFFSET(colors), AV_OPT_TYPE_STRING, {.str = "red|green|blue|yellow|orange|lime|pink|magenta|brown" }, 0, 0, V|F },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(anequalizer);
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static void draw_curves(AVFilterContext *ctx, AVFilterLink *inlink, AVFrame *out)
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{
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AudioNEqualizerContext *s = ctx->priv;
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char *colors, *color, *saveptr = NULL;
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int ch, i, n;
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colors = av_strdup(s->colors);
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if (!colors)
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return;
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memset(out->data[0], 0, s->h * out->linesize[0]);
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for (ch = 0; ch < inlink->channels; ch++) {
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uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff };
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int prev_v = -1;
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double f;
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color = av_strtok(ch == 0 ? colors : NULL, " |", &saveptr);
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if (color)
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av_parse_color(fg, color, -1, ctx);
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for (f = 0; f < s->w; f++) {
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double zr, zi, zr2, zi2;
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double Hr, Hi;
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double Hmag = 1;
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double w;
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int v, y, x;
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w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1);
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zr = cos(w);
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zr2 = zr * zr;
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zi = -sin(w);
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zi2 = zi * zi;
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for (n = 0; n < s->nb_filters; n++) {
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if (s->filters[n].channel != ch ||
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s->filters[n].ignore)
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continue;
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for (i = 0; i < FILTER_ORDER / 2; i++) {
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FoSection *S = &s->filters[n].section[i];
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/* H *= (((((S->b4 * z + S->b3) * z + S->b2) * z + S->b1) * z + S->b0) /
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((((S->a4 * z + S->a3) * z + S->a2) * z + S->a1) * z + S->a0)); */
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Hr = S->b4*(1-8*zr2*zi2) + S->b2*(zr2-zi2) + zr*(S->b1+S->b3*(zr2-3*zi2))+ S->b0;
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Hi = zi*(S->b3*(3*zr2-zi2) + S->b1 + 2*zr*(2*S->b4*(zr2-zi2) + S->b2));
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Hmag *= hypot(Hr, Hi);
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Hr = S->a4*(1-8*zr2*zi2) + S->a2*(zr2-zi2) + zr*(S->a1+S->a3*(zr2-3*zi2))+ S->a0;
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Hi = zi*(S->a3*(3*zr2-zi2) + S->a1 + 2*zr*(2*S->a4*(zr2-zi2) + S->a2));
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Hmag /= hypot(Hr, Hi);
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}
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}
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v = av_clip((1. + -20 * log10(Hmag) / s->mag) * s->h / 2, 0, s->h - 1);
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x = lrint(f);
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if (prev_v == -1)
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prev_v = v;
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if (v <= prev_v) {
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for (y = v; y <= prev_v; y++)
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AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
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} else {
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for (y = prev_v; y <= v; y++)
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AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg));
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}
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prev_v = v;
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}
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}
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av_free(colors);
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}
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static int config_video(AVFilterLink *outlink)
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{
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AVFilterContext *ctx = outlink->src;
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AudioNEqualizerContext *s = ctx->priv;
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AVFilterLink *inlink = ctx->inputs[0];
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AVFrame *out;
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outlink->w = s->w;
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outlink->h = s->h;
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av_frame_free(&s->video);
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s->video = out = ff_get_video_buffer(outlink, outlink->w, outlink->h);
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if (!out)
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return AVERROR(ENOMEM);
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outlink->sample_aspect_ratio = (AVRational){1,1};
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draw_curves(ctx, inlink, out);
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return 0;
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}
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static av_cold int init(AVFilterContext *ctx)
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{
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AudioNEqualizerContext *s = ctx->priv;
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AVFilterPad pad, vpad;
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pad = (AVFilterPad){
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.name = av_strdup("out0"),
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.type = AVMEDIA_TYPE_AUDIO,
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};
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if (!pad.name)
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return AVERROR(ENOMEM);
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if (s->draw_curves) {
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vpad = (AVFilterPad){
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.name = av_strdup("out1"),
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.type = AVMEDIA_TYPE_VIDEO,
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.config_props = config_video,
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};
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if (!vpad.name)
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return AVERROR(ENOMEM);
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}
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ff_insert_outpad(ctx, 0, &pad);
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if (s->draw_curves)
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ff_insert_outpad(ctx, 1, &vpad);
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return 0;
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}
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterLink *inlink = ctx->inputs[0];
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AVFilterLink *outlink = ctx->outputs[0];
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AudioNEqualizerContext *s = ctx->priv;
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AVFilterFormats *formats;
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AVFilterChannelLayouts *layouts;
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static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE };
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static const enum AVSampleFormat sample_fmts[] = {
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AV_SAMPLE_FMT_DBLP,
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AV_SAMPLE_FMT_NONE
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};
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int ret;
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if (s->draw_curves) {
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AVFilterLink *videolink = ctx->outputs[1];
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formats = ff_make_format_list(pix_fmts);
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if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0)
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return ret;
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}
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formats = ff_make_format_list(sample_fmts);
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if ((ret = ff_formats_ref(formats, &inlink->out_formats)) < 0 ||
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(ret = ff_formats_ref(formats, &outlink->in_formats)) < 0)
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return ret;
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layouts = ff_all_channel_counts();
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if ((ret = ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts)) < 0 ||
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(ret = ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts)) < 0)
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return ret;
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formats = ff_all_samplerates();
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if ((ret = ff_formats_ref(formats, &inlink->out_samplerates)) < 0 ||
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(ret = ff_formats_ref(formats, &outlink->in_samplerates)) < 0)
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return ret;
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return 0;
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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AudioNEqualizerContext *s = ctx->priv;
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av_freep(&ctx->output_pads[0].name);
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if (s->draw_curves)
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av_freep(&ctx->output_pads[1].name);
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av_frame_free(&s->video);
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av_freep(&s->filters);
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s->nb_filters = 0;
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s->nb_allocated = 0;
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}
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static void butterworth_fo_section(FoSection *S, double beta,
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double si, double g, double g0,
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double D, double c0)
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{
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if (c0 == 1 || c0 == -1) {
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S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
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S->b1 = 2*c0*(g*g*beta*beta - g0*g0)/D;
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S->b2 = (g*g*beta*beta - 2*g0*g*beta*si + g0*g0)/D;
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S->b3 = 0;
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S->b4 = 0;
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S->a0 = 1;
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S->a1 = 2*c0*(beta*beta - 1)/D;
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S->a2 = (beta*beta - 2*beta*si + 1)/D;
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S->a3 = 0;
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S->a4 = 0;
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} else {
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S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D;
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S->b1 = -4*c0*(g0*g0 + g*g0*si*beta)/D;
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S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - g*g*beta*beta)/D;
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S->b3 = -4*c0*(g0*g0 - g*g0*si*beta)/D;
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S->b4 = (g*g*beta*beta - 2*g*g0*si*beta + g0*g0)/D;
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S->a0 = 1;
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S->a1 = -4*c0*(1 + si*beta)/D;
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S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/D;
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S->a3 = -4*c0*(1 - si*beta)/D;
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S->a4 = (beta*beta - 2*si*beta + 1)/D;
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}
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}
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static void butterworth_bp_filter(EqualizatorFilter *f,
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int N, double w0, double wb,
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double G, double Gb, double G0)
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{
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double g, c0, g0, beta;
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double epsilon;
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int r = N % 2;
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int L = (N - r) / 2;
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int i;
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if (G == 0 && G0 == 0) {
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f->section[0].a0 = 1;
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f->section[0].b0 = 1;
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f->section[1].a0 = 1;
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f->section[1].b0 = 1;
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return;
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}
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G = ff_exp10(G/20);
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Gb = ff_exp10(Gb/20);
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G0 = ff_exp10(G0/20);
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epsilon = sqrt((G * G - Gb * Gb) / (Gb * Gb - G0 * G0));
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g = pow(G, 1.0 / N);
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g0 = pow(G0, 1.0 / N);
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beta = pow(epsilon, -1.0 / N) * tan(wb/2);
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c0 = cos(w0);
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for (i = 1; i <= L; i++) {
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double ui = (2.0 * i - 1) / N;
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double si = sin(M_PI * ui / 2.0);
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double Di = beta * beta + 2 * si * beta + 1;
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butterworth_fo_section(&f->section[i - 1], beta, si, g, g0, Di, c0);
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}
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}
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static void chebyshev1_fo_section(FoSection *S, double a,
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double c, double tetta_b,
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double g0, double si, double b,
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double D, double c0)
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{
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if (c0 == 1 || c0 == -1) {
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S->b0 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) + 2*g0*b*si*tetta_b*tetta_b + g0*g0)/D;
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S->b1 = 2*c0*(tetta_b*tetta_b*(b*b+g0*g0*c*c) - g0*g0)/D;
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S->b2 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) - 2*g0*b*si*tetta_b + g0*g0)/D;
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S->b3 = 0;
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S->b4 = 0;
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S->a0 = 1;
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S->a1 = 2*c0*(tetta_b*tetta_b*(a*a+c*c) - 1)/D;
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S->a2 = (tetta_b*tetta_b*(a*a+c*c) - 2*a*si*tetta_b + 1)/D;
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S->a3 = 0;
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S->a4 = 0;
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} else {
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S->b0 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b + 2*g0*b*si*tetta_b + g0*g0)/D;
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S->b1 = -4*c0*(g0*g0 + g0*b*si*tetta_b)/D;
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S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - (b*b + g0*g0*c*c)*tetta_b*tetta_b)/D;
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S->b3 = -4*c0*(g0*g0 - g0*b*si*tetta_b)/D;
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S->b4 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b - 2*g0*b*si*tetta_b + g0*g0)/D;
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S->a0 = 1;
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S->a1 = -4*c0*(1 + a*si*tetta_b)/D;
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S->a2 = 2*(1 + 2*c0*c0 - (a*a + c*c)*tetta_b*tetta_b)/D;
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S->a3 = -4*c0*(1 - a*si*tetta_b)/D;
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S->a4 = ((a*a + c*c)*tetta_b*tetta_b - 2*a*si*tetta_b + 1)/D;
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}
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}
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static void chebyshev1_bp_filter(EqualizatorFilter *f,
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int N, double w0, double wb,
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double G, double Gb, double G0)
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{
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double a, b, c0, g0, alfa, beta, tetta_b;
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double epsilon;
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int r = N % 2;
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int L = (N - r) / 2;
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int i;
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if (G == 0 && G0 == 0) {
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f->section[0].a0 = 1;
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f->section[0].b0 = 1;
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f->section[1].a0 = 1;
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f->section[1].b0 = 1;
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return;
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}
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G = ff_exp10(G/20);
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Gb = ff_exp10(Gb/20);
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G0 = ff_exp10(G0/20);
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epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
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g0 = pow(G0,1.0/N);
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alfa = pow(1.0/epsilon + sqrt(1 + 1/(epsilon*epsilon)), 1.0/N);
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beta = pow(G/epsilon + Gb * sqrt(1 + 1/(epsilon*epsilon)), 1.0/N);
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a = 0.5 * (alfa - 1.0/alfa);
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b = 0.5 * (beta - g0*g0*(1/beta));
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tetta_b = tan(wb/2);
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c0 = cos(w0);
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for (i = 1; i <= L; i++) {
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double ui = (2.0*i-1.0)/N;
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double ci = cos(M_PI*ui/2.0);
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double si = sin(M_PI*ui/2.0);
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double Di = (a*a + ci*ci)*tetta_b*tetta_b + 2.0*a*si*tetta_b + 1;
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chebyshev1_fo_section(&f->section[i - 1], a, ci, tetta_b, g0, si, b, Di, c0);
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}
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}
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static void chebyshev2_fo_section(FoSection *S, double a,
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double c, double tetta_b,
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double g, double si, double b,
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double D, double c0)
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{
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if (c0 == 1 || c0 == -1) {
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S->b0 = (g*g*tetta_b*tetta_b + 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;
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S->b1 = 2*c0*(g*g*tetta_b*tetta_b - b*b - g*g*c*c)/D;
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S->b2 = (g*g*tetta_b*tetta_b - 2*tetta_b*g*b*si + b*b + g*g*c*c)/D;
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S->b3 = 0;
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S->b4 = 0;
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S->a0 = 1;
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S->a1 = 2*c0*(tetta_b*tetta_b - a*a - c*c)/D;
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|
S->a2 = (tetta_b*tetta_b - 2*tetta_b*a*si + a*a + c*c)/D;
|
|
S->a3 = 0;
|
|
S->a4 = 0;
|
|
} else {
|
|
S->b0 = (g*g*tetta_b*tetta_b + 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;
|
|
S->b1 = -4*c0*(b*b + g*g*c*c + g*b*si*tetta_b)/D;
|
|
S->b2 = 2*((b*b + g*g*c*c)*(1 + 2*c0*c0) - g*g*tetta_b*tetta_b)/D;
|
|
S->b3 = -4*c0*(b*b + g*g*c*c - g*b*si*tetta_b)/D;
|
|
S->b4 = (g*g*tetta_b*tetta_b - 2*g*b*si*tetta_b + b*b + g*g*c*c)/D;
|
|
|
|
S->a0 = 1;
|
|
S->a1 = -4*c0*(a*a + c*c + a*si*tetta_b)/D;
|
|
S->a2 = 2*((a*a + c*c)*(1 + 2*c0*c0) - tetta_b*tetta_b)/D;
|
|
S->a3 = -4*c0*(a*a + c*c - a*si*tetta_b)/D;
|
|
S->a4 = (tetta_b*tetta_b - 2*a*si*tetta_b + a*a + c*c)/D;
|
|
}
|
|
}
|
|
|
|
static void chebyshev2_bp_filter(EqualizatorFilter *f,
|
|
int N, double w0, double wb,
|
|
double G, double Gb, double G0)
|
|
{
|
|
double a, b, c0, tetta_b;
|
|
double epsilon, g, eu, ew;
|
|
int r = N % 2;
|
|
int L = (N - r) / 2;
|
|
int i;
|
|
|
|
if (G == 0 && G0 == 0) {
|
|
f->section[0].a0 = 1;
|
|
f->section[0].b0 = 1;
|
|
f->section[1].a0 = 1;
|
|
f->section[1].b0 = 1;
|
|
return;
|
|
}
|
|
|
|
G = ff_exp10(G/20);
|
|
Gb = ff_exp10(Gb/20);
|
|
G0 = ff_exp10(G0/20);
|
|
|
|
epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0));
|
|
g = pow(G, 1.0 / N);
|
|
eu = pow(epsilon + sqrt(1 + epsilon*epsilon), 1.0/N);
|
|
ew = pow(G0*epsilon + Gb*sqrt(1 + epsilon*epsilon), 1.0/N);
|
|
a = (eu - 1.0/eu)/2.0;
|
|
b = (ew - g*g/ew)/2.0;
|
|
tetta_b = tan(wb/2);
|
|
c0 = cos(w0);
|
|
|
|
for (i = 1; i <= L; i++) {
|
|
double ui = (2.0 * i - 1.0)/N;
|
|
double ci = cos(M_PI * ui / 2.0);
|
|
double si = sin(M_PI * ui / 2.0);
|
|
double Di = tetta_b*tetta_b + 2*a*si*tetta_b + a*a + ci*ci;
|
|
|
|
chebyshev2_fo_section(&f->section[i - 1], a, ci, tetta_b, g, si, b, Di, c0);
|
|
}
|
|
}
|
|
|
|
static double butterworth_compute_bw_gain_db(double gain)
|
|
{
|
|
double bw_gain = 0;
|
|
|
|
if (gain <= -6)
|
|
bw_gain = gain + 3;
|
|
else if(gain > -6 && gain < 6)
|
|
bw_gain = gain * 0.5;
|
|
else if(gain >= 6)
|
|
bw_gain = gain - 3;
|
|
|
|
return bw_gain;
|
|
}
|
|
|
|
static double chebyshev1_compute_bw_gain_db(double gain)
|
|
{
|
|
double bw_gain = 0;
|
|
|
|
if (gain <= -6)
|
|
bw_gain = gain + 1;
|
|
else if(gain > -6 && gain < 6)
|
|
bw_gain = gain * 0.9;
|
|
else if(gain >= 6)
|
|
bw_gain = gain - 1;
|
|
|
|
return bw_gain;
|
|
}
|
|
|
|
static double chebyshev2_compute_bw_gain_db(double gain)
|
|
{
|
|
double bw_gain = 0;
|
|
|
|
if (gain <= -6)
|
|
bw_gain = -3;
|
|
else if(gain > -6 && gain < 6)
|
|
bw_gain = gain * 0.3;
|
|
else if(gain >= 6)
|
|
bw_gain = 3;
|
|
|
|
return bw_gain;
|
|
}
|
|
|
|
static inline double hz_2_rad(double x, double fs)
|
|
{
|
|
return 2 * M_PI * x / fs;
|
|
}
|
|
|
|
static void equalizer(EqualizatorFilter *f, double sample_rate)
|
|
{
|
|
double w0 = hz_2_rad(f->freq, sample_rate);
|
|
double wb = hz_2_rad(f->width, sample_rate);
|
|
double bw_gain;
|
|
|
|
switch (f->type) {
|
|
case BUTTERWORTH:
|
|
bw_gain = butterworth_compute_bw_gain_db(f->gain);
|
|
butterworth_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
|
|
break;
|
|
case CHEBYSHEV1:
|
|
bw_gain = chebyshev1_compute_bw_gain_db(f->gain);
|
|
chebyshev1_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
|
|
break;
|
|
case CHEBYSHEV2:
|
|
bw_gain = chebyshev2_compute_bw_gain_db(f->gain);
|
|
chebyshev2_bp_filter(f, FILTER_ORDER, w0, wb, f->gain, bw_gain, 0);
|
|
break;
|
|
}
|
|
|
|
}
|
|
|
|
static int add_filter(AudioNEqualizerContext *s, AVFilterLink *inlink)
|
|
{
|
|
equalizer(&s->filters[s->nb_filters], inlink->sample_rate);
|
|
if (s->nb_filters >= s->nb_allocated) {
|
|
EqualizatorFilter *filters;
|
|
|
|
filters = av_calloc(s->nb_allocated, 2 * sizeof(*s->filters));
|
|
if (!filters)
|
|
return AVERROR(ENOMEM);
|
|
memcpy(filters, s->filters, sizeof(*s->filters) * s->nb_allocated);
|
|
av_free(s->filters);
|
|
s->filters = filters;
|
|
s->nb_allocated *= 2;
|
|
}
|
|
s->nb_filters++;
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int config_input(AVFilterLink *inlink)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AudioNEqualizerContext *s = ctx->priv;
|
|
char *args = av_strdup(s->args);
|
|
char *saveptr = NULL;
|
|
int ret = 0;
|
|
|
|
if (!args)
|
|
return AVERROR(ENOMEM);
|
|
|
|
s->nb_allocated = 32 * inlink->channels;
|
|
s->filters = av_calloc(inlink->channels, 32 * sizeof(*s->filters));
|
|
if (!s->filters) {
|
|
s->nb_allocated = 0;
|
|
av_free(args);
|
|
return AVERROR(ENOMEM);
|
|
}
|
|
|
|
while (1) {
|
|
char *arg = av_strtok(s->nb_filters == 0 ? args : NULL, "|", &saveptr);
|
|
|
|
if (!arg)
|
|
break;
|
|
|
|
s->filters[s->nb_filters].type = 0;
|
|
if (sscanf(arg, "c%d f=%lf w=%lf g=%lf t=%d", &s->filters[s->nb_filters].channel,
|
|
&s->filters[s->nb_filters].freq,
|
|
&s->filters[s->nb_filters].width,
|
|
&s->filters[s->nb_filters].gain,
|
|
&s->filters[s->nb_filters].type) != 5 &&
|
|
sscanf(arg, "c%d f=%lf w=%lf g=%lf", &s->filters[s->nb_filters].channel,
|
|
&s->filters[s->nb_filters].freq,
|
|
&s->filters[s->nb_filters].width,
|
|
&s->filters[s->nb_filters].gain) != 4 ) {
|
|
av_free(args);
|
|
return AVERROR(EINVAL);
|
|
}
|
|
|
|
if (s->filters[s->nb_filters].freq < 0 ||
|
|
s->filters[s->nb_filters].freq > inlink->sample_rate / 2.0)
|
|
s->filters[s->nb_filters].ignore = 1;
|
|
|
|
if (s->filters[s->nb_filters].channel < 0 ||
|
|
s->filters[s->nb_filters].channel >= inlink->channels)
|
|
s->filters[s->nb_filters].ignore = 1;
|
|
|
|
s->filters[s->nb_filters].type = av_clip(s->filters[s->nb_filters].type, 0, NB_TYPES - 1);
|
|
ret = add_filter(s, inlink);
|
|
if (ret < 0)
|
|
break;
|
|
}
|
|
|
|
av_free(args);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
|
|
char *res, int res_len, int flags)
|
|
{
|
|
AudioNEqualizerContext *s = ctx->priv;
|
|
AVFilterLink *inlink = ctx->inputs[0];
|
|
int ret = AVERROR(ENOSYS);
|
|
|
|
if (!strcmp(cmd, "change")) {
|
|
double freq, width, gain;
|
|
int filter;
|
|
|
|
if (sscanf(args, "%d|f=%lf|w=%lf|g=%lf", &filter, &freq, &width, &gain) != 4)
|
|
return AVERROR(EINVAL);
|
|
|
|
if (filter < 0 || filter >= s->nb_filters)
|
|
return AVERROR(EINVAL);
|
|
|
|
if (freq < 0 || freq > inlink->sample_rate / 2.0)
|
|
return AVERROR(EINVAL);
|
|
|
|
s->filters[filter].freq = freq;
|
|
s->filters[filter].width = width;
|
|
s->filters[filter].gain = gain;
|
|
equalizer(&s->filters[filter], inlink->sample_rate);
|
|
if (s->draw_curves)
|
|
draw_curves(ctx, inlink, s->video);
|
|
|
|
ret = 0;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static inline double section_process(FoSection *S, double in)
|
|
{
|
|
double out;
|
|
|
|
out = S->b0 * in;
|
|
out+= S->b1 * S->num[0] - S->denum[0] * S->a1;
|
|
out+= S->b2 * S->num[1] - S->denum[1] * S->a2;
|
|
out+= S->b3 * S->num[2] - S->denum[2] * S->a3;
|
|
out+= S->b4 * S->num[3] - S->denum[3] * S->a4;
|
|
|
|
S->num[3] = S->num[2];
|
|
S->num[2] = S->num[1];
|
|
S->num[1] = S->num[0];
|
|
S->num[0] = in;
|
|
|
|
S->denum[3] = S->denum[2];
|
|
S->denum[2] = S->denum[1];
|
|
S->denum[1] = S->denum[0];
|
|
S->denum[0] = out;
|
|
|
|
return out;
|
|
}
|
|
|
|
static double process_sample(FoSection *s1, double in)
|
|
{
|
|
double p0 = in, p1;
|
|
int i;
|
|
|
|
for (i = 0; i < FILTER_ORDER / 2; i++) {
|
|
p1 = section_process(&s1[i], p0);
|
|
p0 = p1;
|
|
}
|
|
|
|
return p1;
|
|
}
|
|
|
|
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
|
|
{
|
|
AVFilterContext *ctx = inlink->dst;
|
|
AudioNEqualizerContext *s = ctx->priv;
|
|
AVFilterLink *outlink = ctx->outputs[0];
|
|
double *bptr;
|
|
int i, n;
|
|
|
|
for (i = 0; i < s->nb_filters; i++) {
|
|
EqualizatorFilter *f = &s->filters[i];
|
|
|
|
if (f->gain == 0. || f->ignore)
|
|
continue;
|
|
|
|
bptr = (double *)buf->extended_data[f->channel];
|
|
for (n = 0; n < buf->nb_samples; n++) {
|
|
double sample = bptr[n];
|
|
|
|
sample = process_sample(f->section, sample);
|
|
bptr[n] = sample;
|
|
}
|
|
}
|
|
|
|
if (s->draw_curves) {
|
|
const int64_t pts = buf->pts +
|
|
av_rescale_q(buf->nb_samples, (AVRational){ 1, inlink->sample_rate },
|
|
outlink->time_base);
|
|
int ret;
|
|
|
|
s->video->pts = pts;
|
|
ret = ff_filter_frame(ctx->outputs[1], av_frame_clone(s->video));
|
|
if (ret < 0)
|
|
return ret;
|
|
}
|
|
|
|
return ff_filter_frame(outlink, buf);
|
|
}
|
|
|
|
static const AVFilterPad inputs[] = {
|
|
{
|
|
.name = "default",
|
|
.type = AVMEDIA_TYPE_AUDIO,
|
|
.config_props = config_input,
|
|
.filter_frame = filter_frame,
|
|
.needs_writable = 1,
|
|
},
|
|
{ NULL }
|
|
};
|
|
|
|
AVFilter ff_af_anequalizer = {
|
|
.name = "anequalizer",
|
|
.description = NULL_IF_CONFIG_SMALL("Apply high-order audio parametric multi band equalizer."),
|
|
.priv_size = sizeof(AudioNEqualizerContext),
|
|
.priv_class = &anequalizer_class,
|
|
.init = init,
|
|
.uninit = uninit,
|
|
.query_formats = query_formats,
|
|
.inputs = inputs,
|
|
.outputs = NULL,
|
|
.flags = AVFILTER_FLAG_DYNAMIC_OUTPUTS,
|
|
.process_command = process_command,
|
|
};
|