2 * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
4 * This file is part of FFmpeg.
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 #include "libavutil/channel_layout.h"
22 #include "libavutil/opt.h"
27 typedef struct StereoToolsContext {
45 double phase_sin_coef;
46 double phase_cos_coef;
48 double inv_atan_shape;
57 #define OFFSET(x) offsetof(StereoToolsContext, x)
58 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
60 static const AVOption stereotools_options[] = {
61 { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
62 { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
63 { "balance_in", "set balance in", OFFSET(balance_in), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
64 { "balance_out", "set balance out", OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
65 { "softclip", "enable softclip", OFFSET(softclip), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
66 { "mutel", "mute L", OFFSET(mute_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
67 { "muter", "mute R", OFFSET(mute_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
68 { "phasel", "phase L", OFFSET(phase_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
69 { "phaser", "phase R", OFFSET(phase_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
70 { "mode", "set stereo mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 6, A, "mode" },
71 { "lr>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
72 { "lr>ms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
73 { "ms>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "mode" },
74 { "lr>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "mode" },
75 { "lr>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, A, "mode" },
76 { "lr>l+r", 0, 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, A, "mode" },
77 { "lr>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, A, "mode" },
78 { "slev", "set side level", OFFSET(slev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
79 { "sbal", "set side balance", OFFSET(sbal), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
80 { "mlev", "set middle level", OFFSET(mlev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
81 { "mpan", "set middle pan", OFFSET(mpan), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
82 { "base", "set stereo base", OFFSET(base), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
83 { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20, 20, A },
84 { "sclevel", "set S/C level", OFFSET(sc_level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 100, A },
85 { "phase", "set stereo phase", OFFSET(phase), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 360, A },
89 AVFILTER_DEFINE_CLASS(stereotools);
91 static int query_formats(AVFilterContext *ctx)
93 AVFilterFormats *formats = NULL;
94 AVFilterChannelLayouts *layout = NULL;
97 if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
98 (ret = ff_set_common_formats (ctx , formats )) < 0 ||
99 (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
100 (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
103 formats = ff_all_samplerates();
104 return ff_set_common_samplerates(ctx, formats);
107 static int config_input(AVFilterLink *inlink)
109 AVFilterContext *ctx = inlink->dst;
110 StereoToolsContext *s = ctx->priv;
112 s->length = 2 * inlink->sample_rate * 0.05;
113 if (s->length <= 1 || s->length & 1) {
114 av_log(ctx, AV_LOG_ERROR, "sample rate is too small\n");
115 return AVERROR(EINVAL);
117 s->buffer = av_calloc(s->length, sizeof(*s->buffer));
119 return AVERROR(ENOMEM);
121 s->inv_atan_shape = 1.0 / atan(s->sc_level);
122 s->phase_cos_coef = cos(s->phase / 180 * M_PI);
123 s->phase_sin_coef = sin(s->phase / 180 * M_PI);
128 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
130 AVFilterContext *ctx = inlink->dst;
131 AVFilterLink *outlink = ctx->outputs[0];
132 StereoToolsContext *s = ctx->priv;
133 const double *src = (const double *)in->data[0];
134 const double sb = s->base < 0 ? s->base * 0.5 : s->base;
135 const double sbal = 1 + s->sbal;
136 const double mpan = 1 + s->mpan;
137 const double slev = s->slev;
138 const double mlev = s->mlev;
139 const double balance_in = s->balance_in;
140 const double balance_out = s->balance_out;
141 const double level_in = s->level_in;
142 const double level_out = s->level_out;
143 const double sc_level = s->sc_level;
144 const double delay = s->delay;
145 const int length = s->length;
146 const int mute_l = s->mute_l;
147 const int mute_r = s->mute_r;
148 const int phase_l = s->phase_l;
149 const int phase_r = s->phase_r;
150 double *buffer = s->buffer;
153 int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
157 if (av_frame_is_writable(in)) {
160 out = ff_get_audio_buffer(inlink, in->nb_samples);
163 return AVERROR(ENOMEM);
165 av_frame_copy_props(out, in);
167 dst = (double *)out->data[0];
169 for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
170 double L = src[0], R = src[1], l, r, m, S;
175 L *= 1. - FFMAX(0., balance_in);
176 R *= 1. + FFMIN(0., balance_in);
179 R = s->inv_atan_shape * atan(R * sc_level);
180 L = s->inv_atan_shape * atan(L * sc_level);
187 l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
188 r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
193 l = L * FFMIN(1., 2. - sbal);
194 r = R * FFMIN(1., sbal);
195 L = 0.5 * (l + r) * mlev;
196 R = 0.5 * (l - r) * slev;
199 l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
200 r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
220 l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
221 r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
230 L *= (2. * (1. - phase_l)) - 1.;
231 R *= (2. * (1. - phase_r)) - 1.;
234 buffer[s->pos+1] = R;
237 R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
238 } else if (delay < 0.) {
239 L = buffer[(s->pos - (int)nbuf + length) % length];
242 l = L + sb * L - sb * R;
243 r = R + sb * R - sb * L;
248 l = L * s->phase_cos_coef - R * s->phase_sin_coef;
249 r = L * s->phase_sin_coef + R * s->phase_cos_coef;
254 s->pos = (s->pos + 2) % s->length;
256 L *= 1. - FFMAX(0., balance_out);
257 R *= 1. + FFMIN(0., balance_out);
268 return ff_filter_frame(outlink, out);
271 static av_cold void uninit(AVFilterContext *ctx)
273 StereoToolsContext *s = ctx->priv;
275 av_freep(&s->buffer);
278 static const AVFilterPad inputs[] = {
281 .type = AVMEDIA_TYPE_AUDIO,
282 .filter_frame = filter_frame,
283 .config_props = config_input,
288 static const AVFilterPad outputs[] = {
291 .type = AVMEDIA_TYPE_AUDIO,
296 AVFilter ff_af_stereotools = {
297 .name = "stereotools",
298 .description = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
299 .query_formats = query_formats,
300 .priv_size = sizeof(StereoToolsContext),
301 .priv_class = &stereotools_class,