362 lines
12 KiB
C
362 lines
12 KiB
C
/*
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* Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
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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/channel_layout.h"
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#include "libavutil/opt.h"
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#include "avfilter.h"
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#include "audio.h"
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#include "formats.h"
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typedef struct StereoToolsContext {
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const AVClass *class;
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int softclip;
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int mute_l;
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int mute_r;
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int phase_l;
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int phase_r;
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int mode;
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int bmode_in;
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int bmode_out;
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double slev;
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double sbal;
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double mlev;
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double mpan;
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double phase;
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double base;
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double delay;
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double balance_in;
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double balance_out;
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double phase_sin_coef;
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double phase_cos_coef;
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double sc_level;
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double inv_atan_shape;
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double level_in;
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double level_out;
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double *buffer;
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int length;
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int pos;
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} StereoToolsContext;
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#define OFFSET(x) offsetof(StereoToolsContext, x)
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#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
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static const AVOption stereotools_options[] = {
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{ "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "balance_in", "set balance in", OFFSET(balance_in), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
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{ "balance_out", "set balance out", OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
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{ "softclip", "enable softclip", OFFSET(softclip), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ "mutel", "mute L", OFFSET(mute_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ "muter", "mute R", OFFSET(mute_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ "phasel", "phase L", OFFSET(phase_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ "phaser", "phase R", OFFSET(phase_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
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{ "mode", "set stereo mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 8, A, "mode" },
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{ "lr>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
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{ "lr>ms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
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{ "ms>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "mode" },
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{ "lr>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "mode" },
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{ "lr>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, A, "mode" },
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{ "lr>l+r", 0, 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, A, "mode" },
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{ "lr>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, A, "mode" },
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{ "ms>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, A, "mode" },
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{ "ms>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, A, "mode" },
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{ "slev", "set side level", OFFSET(slev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "sbal", "set side balance", OFFSET(sbal), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
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{ "mlev", "set middle level", OFFSET(mlev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
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{ "mpan", "set middle pan", OFFSET(mpan), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
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{ "base", "set stereo base", OFFSET(base), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
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{ "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20, 20, A },
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{ "sclevel", "set S/C level", OFFSET(sc_level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 100, A },
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{ "phase", "set stereo phase", OFFSET(phase), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 360, A },
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{ "bmode_in", "set balance in mode", OFFSET(bmode_in), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
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{ "balance", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "bmode" },
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{ "amplitude", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "bmode" },
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{ "power", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "bmode" },
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{ "bmode_out", "set balance out mode", OFFSET(bmode_out), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
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{ NULL }
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};
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AVFILTER_DEFINE_CLASS(stereotools);
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static int query_formats(AVFilterContext *ctx)
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{
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AVFilterFormats *formats = NULL;
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AVFilterChannelLayouts *layout = NULL;
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int ret;
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if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
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(ret = ff_set_common_formats (ctx , formats )) < 0 ||
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(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
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(ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
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return ret;
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formats = ff_all_samplerates();
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return ff_set_common_samplerates(ctx, formats);
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}
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static int config_input(AVFilterLink *inlink)
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{
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AVFilterContext *ctx = inlink->dst;
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StereoToolsContext *s = ctx->priv;
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s->length = 2 * inlink->sample_rate * 0.05;
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if (s->length <= 1 || s->length & 1) {
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av_log(ctx, AV_LOG_ERROR, "sample rate is too small\n");
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return AVERROR(EINVAL);
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}
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s->buffer = av_calloc(s->length, sizeof(*s->buffer));
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if (!s->buffer)
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return AVERROR(ENOMEM);
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s->inv_atan_shape = 1.0 / atan(s->sc_level);
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s->phase_cos_coef = cos(s->phase / 180 * M_PI);
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s->phase_sin_coef = sin(s->phase / 180 * M_PI);
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return 0;
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}
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static int filter_frame(AVFilterLink *inlink, AVFrame *in)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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StereoToolsContext *s = ctx->priv;
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const double *src = (const double *)in->data[0];
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const double sb = s->base < 0 ? s->base * 0.5 : s->base;
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const double sbal = 1 + s->sbal;
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const double mpan = 1 + s->mpan;
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const double slev = s->slev;
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const double mlev = s->mlev;
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const double balance_in = s->balance_in;
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const double balance_out = s->balance_out;
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const double level_in = s->level_in;
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const double level_out = s->level_out;
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const double sc_level = s->sc_level;
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const double delay = s->delay;
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const int length = s->length;
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const int mute_l = s->mute_l;
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const int mute_r = s->mute_r;
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const int phase_l = s->phase_l;
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const int phase_r = s->phase_r;
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double *buffer = s->buffer;
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AVFrame *out;
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double *dst;
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int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
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int n;
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nbuf -= nbuf % 2;
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if (av_frame_is_writable(in)) {
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out = in;
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} else {
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out = ff_get_audio_buffer(outlink, in->nb_samples);
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if (!out) {
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av_frame_free(&in);
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return AVERROR(ENOMEM);
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}
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av_frame_copy_props(out, in);
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}
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dst = (double *)out->data[0];
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for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
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double L = src[0], R = src[1], l, r, m, S, gl, gr, gd;
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L *= level_in;
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R *= level_in;
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gl = 1. - FFMAX(0., balance_in);
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gr = 1. + FFMIN(0., balance_in);
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switch (s->bmode_in) {
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case 1:
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gd = gl - gr;
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gl = 1. + gd;
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gr = 1. - gd;
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break;
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case 2:
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if (balance_in < 0.) {
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gr = FFMAX(0.5, gr);
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gl = 1. / gr;
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} else if (balance_in > 0.) {
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gl = FFMAX(0.5, gl);
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gr = 1. / gl;
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}
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break;
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}
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L *= gl;
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R *= gr;
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if (s->softclip) {
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R = s->inv_atan_shape * atan(R * sc_level);
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L = s->inv_atan_shape * atan(L * sc_level);
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}
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switch (s->mode) {
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case 0:
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m = (L + R) * 0.5;
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S = (L - R) * 0.5;
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l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
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r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
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L = l;
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R = r;
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break;
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case 1:
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l = L * FFMIN(1., 2. - sbal);
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r = R * FFMIN(1., sbal);
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L = 0.5 * (l + r) * mlev;
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R = 0.5 * (l - r) * slev;
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break;
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case 2:
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l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
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r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
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L = l;
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R = r;
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break;
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case 3:
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R = L;
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break;
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case 4:
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L = R;
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break;
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case 5:
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L = (L + R) / 2;
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R = L;
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break;
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case 6:
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l = L;
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L = R;
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R = l;
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m = (L + R) * 0.5;
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S = (L - R) * 0.5;
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l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
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r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
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L = l;
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R = r;
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break;
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case 7:
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l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
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L = l;
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R = l;
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break;
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case 8:
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r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
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L = r;
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R = r;
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break;
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}
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L *= 1. - mute_l;
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R *= 1. - mute_r;
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L *= (2. * (1. - phase_l)) - 1.;
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R *= (2. * (1. - phase_r)) - 1.;
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buffer[s->pos ] = L;
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buffer[s->pos+1] = R;
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if (delay > 0.) {
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R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
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} else if (delay < 0.) {
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L = buffer[(s->pos - (int)nbuf + length) % length];
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}
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l = L + sb * L - sb * R;
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r = R + sb * R - sb * L;
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L = l;
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R = r;
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l = L * s->phase_cos_coef - R * s->phase_sin_coef;
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r = L * s->phase_sin_coef + R * s->phase_cos_coef;
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L = l;
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R = r;
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s->pos = (s->pos + 2) % s->length;
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gl = 1. - FFMAX(0., balance_out);
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gr = 1. + FFMIN(0., balance_out);
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switch (s->bmode_out) {
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case 1:
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gd = gl - gr;
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gl = 1. + gd;
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gr = 1. - gd;
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break;
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case 2:
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if (balance_out < 0.) {
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gr = FFMAX(0.5, gr);
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gl = 1. / gr;
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} else if (balance_out > 0.) {
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gl = FFMAX(0.5, gl);
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gr = 1. / gl;
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}
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break;
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}
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L *= gl;
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R *= gr;
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L *= level_out;
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R *= level_out;
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dst[0] = L;
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dst[1] = R;
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}
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if (out != in)
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av_frame_free(&in);
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return ff_filter_frame(outlink, out);
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}
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static av_cold void uninit(AVFilterContext *ctx)
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{
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StereoToolsContext *s = ctx->priv;
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av_freep(&s->buffer);
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}
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static const AVFilterPad inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_frame = filter_frame,
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.config_props = config_input,
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},
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{ NULL }
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};
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static const AVFilterPad outputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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},
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{ NULL }
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};
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AVFilter ff_af_stereotools = {
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.name = "stereotools",
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.description = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
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.query_formats = query_formats,
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.priv_size = sizeof(StereoToolsContext),
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.priv_class = &stereotools_class,
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.uninit = uninit,
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.inputs = inputs,
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.outputs = outputs,
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};
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