/* * Copyright (c) Markus Schmidt and Christian Holschuh * * 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/opt.h" #include "avfilter.h" #include "internal.h" #include "audio.h" typedef struct ChannelParams { double blend_old, drive_old; double rdrive, rbdr, kpa, kpb, kna, knb, ap, an, imr, kc, srct, sq, pwrq; double prev_med, prev_out; double hp[5], lp[5]; double hw[4][2], lw[2][2]; } ChannelParams; typedef struct AExciterContext { const AVClass *class; double level_in; double level_out; double amount; double drive; double blend; double freq; double ceil; int listen; ChannelParams *cp; } AExciterContext; #define OFFSET(x) offsetof(AExciterContext, x) #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM static const AVOption aexciter_options[] = { { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A }, { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A }, { "amount", "set amount", OFFSET(amount), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, A }, { "drive", "set harmonics", OFFSET(drive), AV_OPT_TYPE_DOUBLE, {.dbl=8.5}, 0.1, 10, A }, { "blend", "set blend harmonics", OFFSET(blend), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -10, 10, A }, { "freq", "set scope", OFFSET(freq), AV_OPT_TYPE_DOUBLE, {.dbl=7500}, 2000, 12000, A }, { "ceil", "set ceiling", OFFSET(ceil), AV_OPT_TYPE_DOUBLE, {.dbl=9999}, 9999, 20000, A }, { "listen", "enable listen mode", OFFSET(listen), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A }, { NULL } }; AVFILTER_DEFINE_CLASS(aexciter); static inline double M(double x) { return (fabs(x) > 0.00000001) ? x : 0.0; } static inline double D(double x) { x = fabs(x); return (x > 0.00000001) ? sqrt(x) : 0.0; } static void set_params(ChannelParams *p, double blend, double drive, double srate, double freq, double ceil) { double a0, a1, a2, b0, b1, b2, w0, alpha; p->rdrive = 12.0 / drive; p->rbdr = p->rdrive / (10.5 - blend) * 780.0 / 33.0; p->kpa = D(2.0 * (p->rdrive*p->rdrive) - 1.0) + 1.0; p->kpb = (2.0 - p->kpa) / 2.0; p->ap = ((p->rdrive*p->rdrive) - p->kpa + 1.0) / 2.0; p->kc = p->kpa / D(2.0 * D(2.0 * (p->rdrive*p->rdrive) - 1.0) - 2.0 * p->rdrive*p->rdrive); p->srct = (0.1 * srate) / (0.1 * srate + 1.0); p->sq = p->kc*p->kc + 1.0; p->knb = -1.0 * p->rbdr / D(p->sq); p->kna = 2.0 * p->kc * p->rbdr / D(p->sq); p->an = p->rbdr*p->rbdr / p->sq; p->imr = 2.0 * p->knb + D(2.0 * p->kna + 4.0 * p->an - 1.0); p->pwrq = 2.0 / (p->imr + 1.0); w0 = 2 * M_PI * freq / srate; alpha = sin(w0) / (2. * 0.707); a0 = 1 + alpha; a1 = -2 * cos(w0); a2 = 1 - alpha; b0 = (1 + cos(w0)) / 2; b1 = -(1 + cos(w0)); b2 = (1 + cos(w0)) / 2; p->hp[0] =-a1 / a0; p->hp[1] =-a2 / a0; p->hp[2] = b0 / a0; p->hp[3] = b1 / a0; p->hp[4] = b2 / a0; w0 = 2 * M_PI * ceil / srate; alpha = sin(w0) / (2. * 0.707); a0 = 1 + alpha; a1 = -2 * cos(w0); a2 = 1 - alpha; b0 = (1 - cos(w0)) / 2; b1 = 1 - cos(w0); b2 = (1 - cos(w0)) / 2; p->lp[0] =-a1 / a0; p->lp[1] =-a2 / a0; p->lp[2] = b0 / a0; p->lp[3] = b1 / a0; p->lp[4] = b2 / a0; } static double bprocess(double in, const double *const c, double *w1, double *w2) { double out = c[2] * in + *w1; *w1 = c[3] * in + *w2 + c[0] * out; *w2 = c[4] * in + c[1] * out; return out; } static double distortion_process(AExciterContext *s, ChannelParams *p, double in) { double proc = in, med; proc = bprocess(proc, p->hp, &p->hw[0][0], &p->hw[0][1]); proc = bprocess(proc, p->hp, &p->hw[1][0], &p->hw[1][1]); if (proc >= 0.0) { med = (D(p->ap + proc * (p->kpa - proc)) + p->kpb) * p->pwrq; } else { med = (D(p->an - proc * (p->kna + proc)) + p->knb) * p->pwrq * -1.0; } proc = p->srct * (med - p->prev_med + p->prev_out); p->prev_med = M(med); p->prev_out = M(proc); proc = bprocess(proc, p->hp, &p->hw[2][0], &p->hw[2][1]); proc = bprocess(proc, p->hp, &p->hw[3][0], &p->hw[3][1]); if (s->ceil >= 10000.) { proc = bprocess(proc, p->lp, &p->lw[0][0], &p->lw[0][1]); proc = bprocess(proc, p->lp, &p->lw[1][0], &p->lw[1][1]); } return proc; } static int filter_frame(AVFilterLink *inlink, AVFrame *in) { AVFilterContext *ctx = inlink->dst; AExciterContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; AVFrame *out; const double *src = (const double *)in->data[0]; const double level_in = s->level_in; const double level_out = s->level_out; const double amount = s->amount; const double listen = 1.0 - s->listen; double *dst; if (av_frame_is_writable(in)) { out = in; } else { out = ff_get_audio_buffer(inlink, in->nb_samples); if (!out) { av_frame_free(&in); return AVERROR(ENOMEM); } av_frame_copy_props(out, in); } dst = (double *)out->data[0]; for (int n = 0; n < in->nb_samples; n++) { for (int c = 0; c < inlink->ch_layout.nb_channels; c++) { double sample = src[c] * level_in; sample = distortion_process(s, &s->cp[c], sample); sample = sample * amount + listen * src[c]; sample *= level_out; if (ctx->is_disabled) dst[c] = src[c]; else dst[c] = sample; } src += inlink->ch_layout.nb_channels; dst += inlink->ch_layout.nb_channels; } if (in != out) av_frame_free(&in); return ff_filter_frame(outlink, out); } static av_cold void uninit(AVFilterContext *ctx) { AExciterContext *s = ctx->priv; av_freep(&s->cp); } static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; AExciterContext *s = ctx->priv; if (!s->cp) s->cp = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->cp)); if (!s->cp) return AVERROR(ENOMEM); for (int i = 0; i < inlink->ch_layout.nb_channels; i++) set_params(&s->cp[i], s->blend, s->drive, inlink->sample_rate, s->freq, s->ceil); return 0; } static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags) { AVFilterLink *inlink = ctx->inputs[0]; int ret; ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags); if (ret < 0) return ret; return config_input(inlink); } static const AVFilterPad avfilter_af_aexciter_inputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, .config_props = config_input, .filter_frame = filter_frame, }, }; static const AVFilterPad avfilter_af_aexciter_outputs[] = { { .name = "default", .type = AVMEDIA_TYPE_AUDIO, }, }; const AVFilter ff_af_aexciter = { .name = "aexciter", .description = NULL_IF_CONFIG_SMALL("Enhance high frequency part of audio."), .priv_size = sizeof(AExciterContext), .priv_class = &aexciter_class, .uninit = uninit, FILTER_INPUTS(avfilter_af_aexciter_inputs), FILTER_OUTPUTS(avfilter_af_aexciter_outputs), FILTER_SINGLE_SAMPLEFMT(AV_SAMPLE_FMT_DBL), .process_command = process_command, .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL, };