/* * 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 "libavutil/intreadwrite.h" #include "libavutil/avstring.h" #include "libavutil/ffmath.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 zr, zi, zr2, zi2; double Hr, Hi; double Hmag = 1; double w; int v, y, x; w = M_PI * (s->fscale ? pow(s->w - 1, f / s->w) : f) / (s->w - 1); zr = cos(w); zr2 = zr * zr; zi = -sin(w); zi2 = zi * zi; 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)); */ Hr = S->b4*(1-8*zr2*zi2) + S->b2*(zr2-zi2) + zr*(S->b1+S->b3*(zr2-3*zi2))+ S->b0; Hi = zi*(S->b3*(3*zr2-zi2) + S->b1 + 2*zr*(2*S->b4*(zr2-zi2) + S->b2)); Hmag *= hypot(Hr, Hi); Hr = S->a4*(1-8*zr2*zi2) + S->a2*(zr2-zi2) + zr*(S->a1+S->a3*(zr2-3*zi2))+ S->a0; Hi = zi*(S->a3*(3*zr2-zi2) + S->a1 + 2*zr*(2*S->a4*(zr2-zi2) + S->a2)); Hmag /= hypot(Hr, Hi); } } v = av_clip((1. + -20 * log10(Hmag) / 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; int ret; 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) { av_freep(&pad.name); return AVERROR(ENOMEM); } } ret = ff_insert_outpad(ctx, 0, &pad); if (ret < 0) { av_freep(&pad.name); return ret; } if (s->draw_curves) { ret = ff_insert_outpad(ctx, 1, &vpad); if (ret < 0) { av_freep(&vpad.name); return ret; } } 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; for (int i = 0; i < ctx->nb_outputs; i++) av_freep(&ctx->output_pads[i].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 = 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); 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 = ff_exp10(G/20); Gb = ff_exp10(Gb/20); G0 = ff_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 + 1/(epsilon*epsilon)), 1.0/N); beta = pow(G/epsilon + Gb * sqrt(1 + 1/(epsilon*epsilon)), 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 = 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, };