From 56c7d2b4dad383eb5dd6243a12c98ef65a52a19c Mon Sep 17 00:00:00 2001 From: Paul B Mahol Date: Sun, 20 Dec 2015 09:50:35 +0100 Subject: [PATCH] avfilter: add high-order parametric multiband equalizer filter Signed-off-by: Paul B Mahol --- Changelog | 1 + configure | 1 + doc/filters.texi | 94 +++++ libavfilter/Makefile | 1 + libavfilter/af_anequalizer.c | 751 +++++++++++++++++++++++++++++++++++ libavfilter/allfilters.c | 1 + libavfilter/version.h | 4 +- 7 files changed, 851 insertions(+), 2 deletions(-) create mode 100644 libavfilter/af_anequalizer.c diff --git a/Changelog b/Changelog index 2b46ddb166..bc025cad02 100644 --- a/Changelog +++ b/Changelog @@ -47,6 +47,7 @@ version : - DXVA2-accelerated VP9 decoding - SOFAlizer: virtual binaural acoustics filter - VAAPI VP9 hwaccel +- audio high-order multiband parametric equalizer version 2.8: diff --git a/configure b/configure index 455449a71e..73abf282f6 100755 --- a/configure +++ b/configure @@ -2841,6 +2841,7 @@ unix_protocol_select="network" # filters amovie_filter_deps="avcodec avformat" +anequalizer_filter_deps="cabs cexp" aresample_filter_deps="swresample" ass_filter_deps="libass" asyncts_filter_deps="avresample" diff --git a/doc/filters.texi b/doc/filters.texi index a55cad4988..bf58833b6a 100644 --- a/doc/filters.texi +++ b/doc/filters.texi @@ -992,6 +992,100 @@ stream ends. The default value is 2 seconds. @end table +@section anequalizer + +High-order parametric multiband equalizer for each channel. + +It accepts the following parameters: +@table @option +@item params + +This option string is in format: +"c@var{chn} f=@var{cf} w=@var{w} g=@var{g} t=@var{f} | ..." +Each equalizer band is separated by '|'. + +@table @option +@item chn +Set channel number to which equalization will be applied. +If input doesn't have that channel the entry is ignored. + +@item cf +Set central frequency for band. +If input doesn't have that frequency the entry is ignored. + +@item w +Set band width in hertz. + +@item g +Set band gain in dB. + +@item f +Set filter type for band, optional, can be: + +@table @samp +@item 0 +Butterworth, this is default. + +@item 1 +Chebyshev type 1. + +@item 2 +Chebyshev type 2. +@end table +@end table + +@item curves +With this option activated frequency response of anequalizer is displayed +in video stream. + +@item size +Set video stream size. Only useful if curves option is activated. + +@item mgain +Set max gain that will be displayed. Only useful if curves option is activated. +Setting this to reasonable value allows to display gain which is derived from +neighbour bands which are too close to each other and thus produce higher gain +when both are activated. + +@item fscale +Set frequency scale used to draw frequency response in video output. +Can be linear or logarithmic. Default is logarithmic. + +@item colors +Set color for each channel curve which is going to be displayed in video stream. +This is list of color names separated by space or by '|'. +Unrecognised or missing colors will be replaced by white color. +@end table + +@subsection Examples + +@itemize +@item +Lower gain by 10 of central frequency 200Hz and width 100 Hz +for first 2 channels using Chebyshev type 1 filter: +@example +anequalizer=c0 f=200 w=100 g=-10 t=1|c1 f=200 w=100 g=-10 t=1 +@end example +@end itemize + +@subsection Commands + +This filter supports the following commands: +@table @option +@item change +Alter existing filter parameters. +Syntax for the commands is : "@var{fN}|f=@var{freq}|w=@var{width}|g=@var{gain}" + +@var{fN} is existing filter number, starting from 0, if no such filter is available +error is returned. +@var{freq} set new frequency parameter. +@var{width} set new width parameter in herz. +@var{gain} set new gain parameter in dB. + +Full filter invocation with asendcmd may look like this: +asendcmd=c='4.0 anequalizer change 0|f=200|w=50|g=1',anequalizer=... +@end table + @section anull Pass the audio source unchanged to the output. diff --git a/libavfilter/Makefile b/libavfilter/Makefile index dea012aa93..adbbc3921b 100644 --- a/libavfilter/Makefile +++ b/libavfilter/Makefile @@ -29,6 +29,7 @@ OBJS-$(CONFIG_ACROSSFADE_FILTER) += af_afade.o OBJS-$(CONFIG_ADELAY_FILTER) += af_adelay.o OBJS-$(CONFIG_AECHO_FILTER) += af_aecho.o OBJS-$(CONFIG_AEMPHASIS_FILTER) += af_aemphasis.o +OBJS-$(CONFIG_ANEQUALIZER_FILTER) += af_anequalizer.o OBJS-$(CONFIG_AEVAL_FILTER) += aeval.o OBJS-$(CONFIG_AFADE_FILTER) += af_afade.o OBJS-$(CONFIG_AFORMAT_FILTER) += af_aformat.o diff --git a/libavfilter/af_anequalizer.c b/libavfilter/af_anequalizer.c new file mode 100644 index 0000000000..a437e2b9dc --- /dev/null +++ b/libavfilter/af_anequalizer.c @@ -0,0 +1,751 @@ +/* + * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen and others + * Copyright (c) 2015 Paul B Mahol + * + * 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 + */ + +#include + +#include "libavutil/intreadwrite.h" +#include "libavutil/avstring.h" +#include "libavutil/opt.h" +#include "libavutil/parseutils.h" +#include "avfilter.h" +#include "internal.h" +#include "audio.h" + +#define FILTER_ORDER 4 + +enum FilterType { + BUTTERWORTH, + CHEBYSHEV1, + CHEBYSHEV2, + NB_TYPES +}; + +typedef struct FoSection { + double a0, a1, a2, a3, a4; + double b0, b1, b2, b3, b4; + + double num[4]; + double denum[4]; +} FoSection; + +typedef struct EqualizatorFilter { + int ignore; + int channel; + int type; + + double freq; + double gain; + double width; + + FoSection section[2]; +} EqualizatorFilter; + +typedef struct AudioNEqualizerContext { + const AVClass *class; + char *args; + char *colors; + int draw_curves; + int w, h; + + double mag; + int fscale; + int nb_filters; + int nb_allocated; + EqualizatorFilter *filters; + AVFrame *video; +} AudioNEqualizerContext; + +#define OFFSET(x) offsetof(AudioNEqualizerContext, x) +#define A AV_OPT_FLAG_AUDIO_PARAM +#define V AV_OPT_FLAG_VIDEO_PARAM +#define F AV_OPT_FLAG_FILTERING_PARAM + +static const AVOption anequalizer_options[] = { + { "params", NULL, OFFSET(args), AV_OPT_TYPE_STRING, {.str=""}, 0, 0, A|F }, + { "curves", "draw frequency response curves", OFFSET(draw_curves), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, V|F }, + { "size", "set video size", OFFSET(w), AV_OPT_TYPE_IMAGE_SIZE, {.str = "hd720"}, 0, 0, V|F }, + { "mgain", "set max gain", OFFSET(mag), AV_OPT_TYPE_DOUBLE, {.dbl=60}, -900, 900, V|F }, + { "fscale", "set frequency scale", OFFSET(fscale), AV_OPT_TYPE_INT, {.i64=1}, 0, 1, V|F, "fscale" }, + { "lin", "linear", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, V|F, "fscale" }, + { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, V|F, "fscale" }, + { "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 }, + { NULL } +}; + +AVFILTER_DEFINE_CLASS(anequalizer); + +static void draw_curves(AVFilterContext *ctx, AVFilterLink *inlink, AVFrame *out) +{ + AudioNEqualizerContext *s = ctx->priv; + char *colors, *color, *saveptr = NULL; + int ch, i, n; + + colors = av_strdup(s->colors); + if (!colors) + return; + + memset(out->data[0], 0, s->h * out->linesize[0]); + + for (ch = 0; ch < inlink->channels; ch++) { + uint8_t fg[4] = { 0xff, 0xff, 0xff, 0xff }; + int prev_v = -1; + double f; + + color = av_strtok(ch == 0 ? colors : NULL, " |", &saveptr); + if (color) + av_parse_color(fg, color, -1, ctx); + + for (f = 0; f < s->w; f++) { + double complex z; + double complex H = 1; + double w; + int v, y, x; + + w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1); + z = 1. / cexp(I * w); + + for (n = 0; n < s->nb_filters; n++) { + if (s->filters[n].channel != ch || + s->filters[n].ignore) + continue; + + for (i = 0; i < FILTER_ORDER / 2; i++) { + FoSection *S = &s->filters[n].section[i]; + + H *= (((((S->b4 * z + S->b3) * z + S->b2) * z + S->b1) * z + S->b0) / + ((((S->a4 * z + S->a3) * z + S->a2) * z + S->a1) * z + S->a0)); + } + } + + v = av_clip((1. + -20 * log10(cabs(H)) / s->mag) * s->h / 2, 0, s->h - 1); + x = lrint(f); + if (prev_v == -1) + prev_v = v; + if (v <= prev_v) { + for (y = v; y <= prev_v; y++) + AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg)); + } else { + for (y = prev_v; y <= v; y++) + AV_WL32(out->data[0] + y * out->linesize[0] + x * 4, AV_RL32(fg)); + } + + prev_v = v; + } + } + + av_free(colors); +} + +static int config_video(AVFilterLink *outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioNEqualizerContext *s = ctx->priv; + AVFilterLink *inlink = ctx->inputs[0]; + AVFrame *out; + + outlink->w = s->w; + outlink->h = s->h; + + av_frame_free(&s->video); + s->video = out = ff_get_video_buffer(outlink, outlink->w, outlink->h); + if (!out) + return AVERROR(ENOMEM); + outlink->sample_aspect_ratio = (AVRational){1,1}; + + draw_curves(ctx, inlink, out); + + return 0; +} + +static av_cold int init(AVFilterContext *ctx) +{ + AudioNEqualizerContext *s = ctx->priv; + AVFilterPad pad, vpad; + + pad = (AVFilterPad){ + .name = av_strdup("out0"), + .type = AVMEDIA_TYPE_AUDIO, + }; + + if (!pad.name) + return AVERROR(ENOMEM); + + if (s->draw_curves) { + vpad = (AVFilterPad){ + .name = av_strdup("out1"), + .type = AVMEDIA_TYPE_VIDEO, + .config_props = config_video, + }; + if (!vpad.name) + return AVERROR(ENOMEM); + } + + ff_insert_outpad(ctx, 0, &pad); + + if (s->draw_curves) + ff_insert_outpad(ctx, 1, &vpad); + + return 0; +} + +static int query_formats(AVFilterContext *ctx) +{ + AVFilterLink *inlink = ctx->inputs[0]; + AVFilterLink *outlink = ctx->outputs[0]; + AudioNEqualizerContext *s = ctx->priv; + AVFilterFormats *formats; + AVFilterChannelLayouts *layouts; + static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGBA, AV_PIX_FMT_NONE }; + static const enum AVSampleFormat sample_fmts[] = { + AV_SAMPLE_FMT_DBLP, + AV_SAMPLE_FMT_NONE + }; + int ret; + + if (s->draw_curves) { + AVFilterLink *videolink = ctx->outputs[1]; + formats = ff_make_format_list(pix_fmts); + if ((ret = ff_formats_ref(formats, &videolink->in_formats)) < 0) + return ret; + } + + formats = ff_make_format_list(sample_fmts); + if ((ret = ff_formats_ref(formats, &inlink->out_formats)) < 0 || + (ret = ff_formats_ref(formats, &outlink->in_formats)) < 0) + return ret; + + layouts = ff_all_channel_counts(); + if ((ret = ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts)) < 0 || + (ret = ff_channel_layouts_ref(layouts, &outlink->in_channel_layouts)) < 0) + return ret; + + formats = ff_all_samplerates(); + if ((ret = ff_formats_ref(formats, &inlink->out_samplerates)) < 0 || + (ret = ff_formats_ref(formats, &outlink->in_samplerates)) < 0) + return ret; + + return 0; +} + +static av_cold void uninit(AVFilterContext *ctx) +{ + AudioNEqualizerContext *s = ctx->priv; + + av_freep(&ctx->output_pads[0].name); + if (s->draw_curves) + av_freep(&ctx->output_pads[1].name); + av_frame_free(&s->video); + av_freep(&s->filters); + s->nb_filters = 0; + s->nb_allocated = 0; +} + +static void butterworth_fo_section(FoSection *S, double beta, + double si, double g, double g0, + double D, double c0) +{ + if (c0 == 1 || c0 == -1) { + S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D; + S->b1 = 2*c0*(g*g*beta*beta - g0*g0)/D; + S->b2 = (g*g*beta*beta - 2*g0*g*beta*si + g0*g0)/D; + S->b3 = 0; + S->b4 = 0; + + S->a0 = 1; + S->a1 = 2*c0*(beta*beta - 1)/D; + S->a2 = (beta*beta - 2*beta*si + 1)/D; + S->a3 = 0; + S->a4 = 0; + } else { + S->b0 = (g*g*beta*beta + 2*g*g0*si*beta + g0*g0)/D; + S->b1 = -4*c0*(g0*g0 + g*g0*si*beta)/D; + S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - g*g*beta*beta)/D; + S->b3 = -4*c0*(g0*g0 - g*g0*si*beta)/D; + S->b4 = (g*g*beta*beta - 2*g*g0*si*beta + g0*g0)/D; + + S->a0 = 1; + S->a1 = -4*c0*(1 + si*beta)/D; + S->a2 = 2*(1 + 2*c0*c0 - beta*beta)/D; + S->a3 = -4*c0*(1 - si*beta)/D; + S->a4 = (beta*beta - 2*si*beta + 1)/D; + } +} + +static void butterworth_bp_filter(EqualizatorFilter *f, + int N, double w0, double wb, + double G, double Gb, double G0) +{ + double g, c0, g0, beta; + double epsilon; + 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 = exp10( G/20); + Gb = exp10(Gb/20); + G0 = exp10(G0/20); + + epsilon = sqrt((G * G - Gb * Gb) / (Gb * Gb - G0 * G0)); + g = pow(G, 1.0 / N); + g0 = pow(G0, 1.0 / N); + beta = pow(epsilon, -1.0 / N) * tan(wb/2); + c0 = cos(w0); + + for (i = 1; i <= L; i++) { + double ui = (2.0 * i - 1) / N; + double si = sin(M_PI * ui / 2.0); + double Di = beta * beta + 2 * si * beta + 1; + + butterworth_fo_section(&f->section[i - 1], beta, si, g, g0, Di, c0); + } +} + +static void chebyshev1_fo_section(FoSection *S, double a, + double c, double tetta_b, + double g0, double si, double b, + double D, double c0) +{ + if (c0 == 1 || c0 == -1) { + S->b0 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) + 2*g0*b*si*tetta_b*tetta_b + g0*g0)/D; + S->b1 = 2*c0*(tetta_b*tetta_b*(b*b+g0*g0*c*c) - g0*g0)/D; + S->b2 = (tetta_b*tetta_b*(b*b+g0*g0*c*c) - 2*g0*b*si*tetta_b + g0*g0)/D; + S->b3 = 0; + S->b4 = 0; + + S->a0 = 1; + S->a1 = 2*c0*(tetta_b*tetta_b*(a*a+c*c) - 1)/D; + S->a2 = (tetta_b*tetta_b*(a*a+c*c) - 2*a*si*tetta_b + 1)/D; + S->a3 = 0; + S->a4 = 0; + } else { + S->b0 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b + 2*g0*b*si*tetta_b + g0*g0)/D; + S->b1 = -4*c0*(g0*g0 + g0*b*si*tetta_b)/D; + S->b2 = 2*(g0*g0*(1 + 2*c0*c0) - (b*b + g0*g0*c*c)*tetta_b*tetta_b)/D; + S->b3 = -4*c0*(g0*g0 - g0*b*si*tetta_b)/D; + S->b4 = ((b*b + g0*g0*c*c)*tetta_b*tetta_b - 2*g0*b*si*tetta_b + g0*g0)/D; + + S->a0 = 1; + S->a1 = -4*c0*(1 + a*si*tetta_b)/D; + S->a2 = 2*(1 + 2*c0*c0 - (a*a + c*c)*tetta_b*tetta_b)/D; + S->a3 = -4*c0*(1 - a*si*tetta_b)/D; + S->a4 = ((a*a + c*c)*tetta_b*tetta_b - 2*a*si*tetta_b + 1)/D; + } +} + +static void chebyshev1_bp_filter(EqualizatorFilter *f, + int N, double w0, double wb, + double G, double Gb, double G0) +{ + double a, b, c0, g0, alfa, beta, tetta_b; + double epsilon; + 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 = exp10( G/20); + Gb = exp10(Gb/20); + G0 = exp10(G0/20); + + epsilon = sqrt((G*G - Gb*Gb) / (Gb*Gb - G0*G0)); + g0 = pow(G0,1.0/N); + alfa = pow(1.0/epsilon + sqrt(1 + pow(epsilon,-2.0)), 1.0/N); + beta = pow(G/epsilon + Gb * sqrt(1 + pow(epsilon,-2.0)), 1.0/N); + a = 0.5 * (alfa - 1.0/alfa); + b = 0.5 * (beta - g0*g0*(1/beta)); + 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 = (a*a + ci*ci)*tetta_b*tetta_b + 2.0*a*si*tetta_b + 1; + + chebyshev1_fo_section(&f->section[i - 1], a, ci, tetta_b, g0, si, b, Di, c0); + } +} + +static void chebyshev2_fo_section(FoSection *S, double a, + double c, double tetta_b, + double g, double si, double b, + double D, double c0) +{ + if (c0 == 1 || c0 == -1) { + S->b0 = (g*g*tetta_b*tetta_b + 2*tetta_b*g*b*si + b*b + g*g*c*c)/D; + S->b1 = 2*c0*(g*g*tetta_b*tetta_b - b*b - g*g*c*c)/D; + S->b2 = (g*g*tetta_b*tetta_b - 2*tetta_b*g*b*si + b*b + g*g*c*c)/D; + S->b3 = 0; + S->b4 = 0; + + S->a0 = 1; + S->a1 = 2*c0*(tetta_b*tetta_b - a*a - c*c)/D; + 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 = exp10( G/20); + Gb = exp10(Gb/20); + G0 = 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; + 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) + 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; + + 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) + 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, +}; diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c index 131e067aef..a039a39922 100644 --- a/libavfilter/allfilters.c +++ b/libavfilter/allfilters.c @@ -59,6 +59,7 @@ void avfilter_register_all(void) REGISTER_FILTER(ALLPASS, allpass, af); REGISTER_FILTER(AMERGE, amerge, af); REGISTER_FILTER(AMIX, amix, af); + REGISTER_FILTER(ANEQUALIZER, anequalizer, af); REGISTER_FILTER(ANULL, anull, af); REGISTER_FILTER(APAD, apad, af); REGISTER_FILTER(APERMS, aperms, af); diff --git a/libavfilter/version.h b/libavfilter/version.h index 79a1f01514..d0c8d8b7be 100644 --- a/libavfilter/version.h +++ b/libavfilter/version.h @@ -30,8 +30,8 @@ #include "libavutil/version.h" #define LIBAVFILTER_VERSION_MAJOR 6 -#define LIBAVFILTER_VERSION_MINOR 21 -#define LIBAVFILTER_VERSION_MICRO 101 +#define LIBAVFILTER_VERSION_MINOR 22 +#define LIBAVFILTER_VERSION_MICRO 100 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \ LIBAVFILTER_VERSION_MINOR, \