aboutsummaryrefslogtreecommitdiffstats
path: root/libavfilter/af_stereotools.c
blob: 8ab184df11c01715881461777375fecba667fe2a (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
/*
 * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
 *
 * 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/channel_layout.h"
#include "libavutil/opt.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"

typedef struct StereoToolsContext {
    const AVClass *class;

    int softclip;
    int mute_l;
    int mute_r;
    int phase_l;
    int phase_r;
    int mode;
    double slev;
    double sbal;
    double mlev;
    double mpan;
    double phase;
    double base;
    double delay;
    double balance_in;
    double balance_out;
    double phase_sin_coef;
    double phase_cos_coef;
    double sc_level;
    double inv_atan_shape;
    double level_in;
    double level_out;

    double *buffer;
    int length;
    int pos;
} StereoToolsContext;

#define OFFSET(x) offsetof(StereoToolsContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption stereotools_options[] = {
    { "level_in",    "set level in",     OFFSET(level_in),    AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
    { "level_out",   "set level out",    OFFSET(level_out),   AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
    { "balance_in",  "set balance in",   OFFSET(balance_in),  AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
    { "balance_out", "set balance out",  OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
    { "softclip",    "enable softclip",  OFFSET(softclip),    AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
    { "mutel",       "mute L",           OFFSET(mute_l),      AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
    { "muter",       "mute R",           OFFSET(mute_r),      AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
    { "phasel",      "phase L",          OFFSET(phase_l),     AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
    { "phaser",      "phase R",          OFFSET(phase_r),     AV_OPT_TYPE_BOOL,   {.i64=0},   0,          1, A },
    { "mode",        "set stereo mode",  OFFSET(mode),        AV_OPT_TYPE_INT,    {.i64=0},   0,          6, A, "mode" },
    {     "lr>lr",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=0},   0,          0, A, "mode" },
    {     "lr>ms",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=1},   0,          0, A, "mode" },
    {     "ms>lr",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=2},   0,          0, A, "mode" },
    {     "lr>ll",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=3},   0,          0, A, "mode" },
    {     "lr>rr",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=4},   0,          0, A, "mode" },
    {     "lr>l+r",  0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=5},   0,          0, A, "mode" },
    {     "lr>rl",   0,                  0,                   AV_OPT_TYPE_CONST,  {.i64=6},   0,          0, A, "mode" },
    { "slev",        "set side level",   OFFSET(slev),        AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
    { "sbal",        "set side balance", OFFSET(sbal),        AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
    { "mlev",        "set middle level", OFFSET(mlev),        AV_OPT_TYPE_DOUBLE, {.dbl=1},   0.015625,  64, A },
    { "mpan",        "set middle pan",   OFFSET(mpan),        AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
    { "base",        "set stereo base",  OFFSET(base),        AV_OPT_TYPE_DOUBLE, {.dbl=0},  -1,          1, A },
    { "delay",       "set delay",        OFFSET(delay),       AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20,         20, A },
    { "sclevel",     "set S/C level",    OFFSET(sc_level),    AV_OPT_TYPE_DOUBLE, {.dbl=1},   1,        100, A },
    { "phase",       "set stereo phase", OFFSET(phase),       AV_OPT_TYPE_DOUBLE, {.dbl=0},   0,        360, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(stereotools);

static int query_formats(AVFilterContext *ctx)
{
    AVFilterFormats *formats = NULL;
    AVFilterChannelLayouts *layout = NULL;
    int ret;

    if ((ret = ff_add_format                 (&formats, AV_SAMPLE_FMT_DBL  )) < 0 ||
        (ret = ff_set_common_formats         (ctx     , formats            )) < 0 ||
        (ret = ff_add_channel_layout         (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
        (ret = ff_set_common_channel_layouts (ctx     , layout             )) < 0)
        return ret;

    formats = ff_all_samplerates();
    return ff_set_common_samplerates(ctx, formats);
}

static int config_input(AVFilterLink *inlink)
{
    AVFilterContext *ctx = inlink->dst;
    StereoToolsContext *s = ctx->priv;

    s->length = 2 * inlink->sample_rate * 0.05;
    if (s->length <= 1 || s->length & 1) {
        av_log(ctx, AV_LOG_ERROR, "sample rate is too small\n");
        return AVERROR(EINVAL);
    }
    s->buffer = av_calloc(s->length, sizeof(*s->buffer));
    if (!s->buffer)
        return AVERROR(ENOMEM);

    s->inv_atan_shape = 1.0 / atan(s->sc_level);
    s->phase_cos_coef = cos(s->phase / 180 * M_PI);
    s->phase_sin_coef = sin(s->phase / 180 * M_PI);

    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AVFilterLink *outlink = ctx->outputs[0];
    StereoToolsContext *s = ctx->priv;
    const double *src = (const double *)in->data[0];
    const double sb = s->base < 0 ? s->base * 0.5 : s->base;
    const double sbal = 1 + s->sbal;
    const double mpan = 1 + s->mpan;
    const double slev = s->slev;
    const double mlev = s->mlev;
    const double balance_in = s->balance_in;
    const double balance_out = s->balance_out;
    const double level_in = s->level_in;
    const double level_out = s->level_out;
    const double sc_level = s->sc_level;
    const double delay = s->delay;
    const int length = s->length;
    const int mute_l = s->mute_l;
    const int mute_r = s->mute_r;
    const int phase_l = s->phase_l;
    const int phase_r = s->phase_r;
    double *buffer = s->buffer;
    AVFrame *out;
    double *dst;
    int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
    int n;

    nbuf -= nbuf % 2;
    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 (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
        double L = src[0], R = src[1], l, r, m, S;

        L *= level_in;
        R *= level_in;

        L *= 1. - FFMAX(0., balance_in);
        R *= 1. + FFMIN(0., balance_in);

        if (s->softclip) {
            R = s->inv_atan_shape * atan(R * sc_level);
            L = s->inv_atan_shape * atan(L * sc_level);
        }

        switch (s->mode) {
        case 0:
            m = (L + R) * 0.5;
            S = (L - R) * 0.5;
            l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
            r = m * mlev * FFMIN(1., mpan)      - S * slev * FFMIN(1., sbal);
            L = l;
            R = r;
            break;
        case 1:
            l = L * FFMIN(1., 2. - sbal);
            r = R * FFMIN(1., sbal);
            L = 0.5 * (l + r) * mlev;
            R = 0.5 * (l - r) * slev;
            break;
        case 2:
            l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
            r = L * mlev * FFMIN(1., mpan)      - R * slev * FFMIN(1., sbal);
            L = l;
            R = r;
            break;
        case 3:
            R = L;
            break;
        case 4:
            L = R;
            break;
        case 5:
            L = (L + R) / 2;
            R = L;
            break;
        case 6:
            l = L;
            L = R;
            R = l;
            m = (L + R) * 0.5;
            S = (L - R) * 0.5;
            l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
            r = m * mlev * FFMIN(1., mpan)      - S * slev * FFMIN(1., sbal);
            L = l;
            R = r;
            break;
        }

        L *= 1. - mute_l;
        R *= 1. - mute_r;

        L *= (2. * (1. - phase_l)) - 1.;
        R *= (2. * (1. - phase_r)) - 1.;

        buffer[s->pos  ] = L;
        buffer[s->pos+1] = R;

        if (delay > 0.) {
            R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
        } else if (delay < 0.) {
            L = buffer[(s->pos - (int)nbuf + length)     % length];
        }

        l = L + sb * L - sb * R;
        r = R + sb * R - sb * L;

        L = l;
        R = r;

        l = L * s->phase_cos_coef - R * s->phase_sin_coef;
        r = L * s->phase_sin_coef + R * s->phase_cos_coef;

        L = l;
        R = r;

        s->pos = (s->pos + 2) % s->length;

        L *= 1. - FFMAX(0., balance_out);
        R *= 1. + FFMIN(0., balance_out);

        L *= level_out;
        R *= level_out;

        dst[0] = L;
        dst[1] = R;
    }

    if (out != in)
        av_frame_free(&in);
    return ff_filter_frame(outlink, out);
}

static av_cold void uninit(AVFilterContext *ctx)
{
    StereoToolsContext *s = ctx->priv;

    av_freep(&s->buffer);
}

static const AVFilterPad inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .filter_frame = filter_frame,
        .config_props = config_input,
    },
    { NULL }
};

static const AVFilterPad outputs[] = {
    {
        .name = "default",
        .type = AVMEDIA_TYPE_AUDIO,
    },
    { NULL }
};

AVFilter ff_af_stereotools = {
    .name           = "stereotools",
    .description    = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
    .query_formats  = query_formats,
    .priv_size      = sizeof(StereoToolsContext),
    .priv_class     = &stereotools_class,
    .uninit         = uninit,
    .inputs         = inputs,
    .outputs        = outputs,
};