GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_asoftclip.c Lines: 0 251 0.0 %
Date: 2021-04-22 14:24:15 Branches: 0 131 0.0 %

Line Branch Exec Source
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/*
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 * Copyright (c) 2019 The FFmpeg Project
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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
19
 */
20
21
#include "libavutil/avassert.h"
22
#include "libavutil/channel_layout.h"
23
#include "libavutil/opt.h"
24
#include "libswresample/swresample.h"
25
#include "avfilter.h"
26
#include "audio.h"
27
#include "formats.h"
28
29
enum ASoftClipTypes {
30
    ASC_HARD = -1,
31
    ASC_TANH,
32
    ASC_ATAN,
33
    ASC_CUBIC,
34
    ASC_EXP,
35
    ASC_ALG,
36
    ASC_QUINTIC,
37
    ASC_SIN,
38
    ASC_ERF,
39
    NB_TYPES,
40
};
41
42
typedef struct ASoftClipContext {
43
    const AVClass *class;
44
45
    int type;
46
    int oversample;
47
    int64_t delay;
48
    double threshold;
49
    double output;
50
    double param;
51
52
    SwrContext *up_ctx;
53
    SwrContext *down_ctx;
54
55
    AVFrame *frame;
56
57
    void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
58
                   int nb_samples, int channels, int start, int end);
59
} ASoftClipContext;
60
61
#define OFFSET(x) offsetof(ASoftClipContext, x)
62
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
63
#define F AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
64
65
static const AVOption asoftclip_options[] = {
66
    { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT,    {.i64=0},         -1, NB_TYPES-1, A, "types" },
67
    { "hard",                NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_HARD},   0,          0, A, "types" },
68
    { "tanh",                NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_TANH},   0,          0, A, "types" },
69
    { "atan",                NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_ATAN},   0,          0, A, "types" },
70
    { "cubic",               NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_CUBIC},  0,          0, A, "types" },
71
    { "exp",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_EXP},    0,          0, A, "types" },
72
    { "alg",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_ALG},    0,          0, A, "types" },
73
    { "quintic",             NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_QUINTIC},0,          0, A, "types" },
74
    { "sin",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_SIN},    0,          0, A, "types" },
75
    { "erf",                 NULL,            0, AV_OPT_TYPE_CONST,  {.i64=ASC_ERF},    0,          0, A, "types" },
76
    { "threshold", "set softclip threshold", OFFSET(threshold), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 1, A },
77
    { "output", "set softclip output gain", OFFSET(output), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.000001, 16, A },
78
    { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01,        3, A },
79
    { "oversample", "set oversample factor", OFFSET(oversample), AV_OPT_TYPE_INT, {.i64=1}, 1, 32, F },
80
    { NULL }
81
};
82
83
AVFILTER_DEFINE_CLASS(asoftclip);
84
85
static int query_formats(AVFilterContext *ctx)
86
{
87
    AVFilterFormats *formats = NULL;
88
    AVFilterChannelLayouts *layouts = NULL;
89
    static const enum AVSampleFormat sample_fmts[] = {
90
        AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
91
        AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
92
        AV_SAMPLE_FMT_NONE
93
    };
94
    int ret;
95
96
    formats = ff_make_format_list(sample_fmts);
97
    if (!formats)
98
        return AVERROR(ENOMEM);
99
    ret = ff_set_common_formats(ctx, formats);
100
    if (ret < 0)
101
        return ret;
102
103
    layouts = ff_all_channel_counts();
104
    if (!layouts)
105
        return AVERROR(ENOMEM);
106
107
    ret = ff_set_common_channel_layouts(ctx, layouts);
108
    if (ret < 0)
109
        return ret;
110
111
    formats = ff_all_samplerates();
112
    return ff_set_common_samplerates(ctx, formats);
113
}
114
115
static void filter_flt(ASoftClipContext *s,
116
                       void **dptr, const void **sptr,
117
                       int nb_samples, int channels,
118
                       int start, int end)
119
{
120
    float threshold = s->threshold;
121
    float gain = s->output * threshold;
122
    float factor = 1.f / threshold;
123
    float param = s->param;
124
125
    for (int c = start; c < end; c++) {
126
        const float *src = sptr[c];
127
        float *dst = dptr[c];
128
129
        switch (s->type) {
130
        case ASC_HARD:
131
            for (int n = 0; n < nb_samples; n++) {
132
                dst[n] = av_clipf(src[n] * factor, -1.f, 1.f);
133
                dst[n] *= gain;
134
            }
135
            break;
136
        case ASC_TANH:
137
            for (int n = 0; n < nb_samples; n++) {
138
                dst[n] = tanhf(src[n] * factor * param);
139
                dst[n] *= gain;
140
            }
141
            break;
142
        case ASC_ATAN:
143
            for (int n = 0; n < nb_samples; n++) {
144
                dst[n] = 2.f / M_PI * atanf(src[n] * factor * param);
145
                dst[n] *= gain;
146
            }
147
            break;
148
        case ASC_CUBIC:
149
            for (int n = 0; n < nb_samples; n++) {
150
                float sample = src[n] * factor;
151
152
                if (FFABS(sample) >= 1.5f)
153
                    dst[n] = FFSIGN(sample);
154
                else
155
                    dst[n] = sample - 0.1481f * powf(sample, 3.f);
156
                dst[n] *= gain;
157
            }
158
            break;
159
        case ASC_EXP:
160
            for (int n = 0; n < nb_samples; n++) {
161
                dst[n] = 2.f / (1.f + expf(-2.f * src[n] * factor)) - 1.;
162
                dst[n] *= gain;
163
            }
164
            break;
165
        case ASC_ALG:
166
            for (int n = 0; n < nb_samples; n++) {
167
                float sample = src[n] * factor;
168
169
                dst[n] = sample / (sqrtf(param + sample * sample));
170
                dst[n] *= gain;
171
            }
172
            break;
173
        case ASC_QUINTIC:
174
            for (int n = 0; n < nb_samples; n++) {
175
                float sample = src[n] * factor;
176
177
                if (FFABS(sample) >= 1.25)
178
                    dst[n] = FFSIGN(sample);
179
                else
180
                    dst[n] = sample - 0.08192f * powf(sample, 5.f);
181
                dst[n] *= gain;
182
            }
183
            break;
184
        case ASC_SIN:
185
            for (int n = 0; n < nb_samples; n++) {
186
                float sample = src[n] * factor;
187
188
                if (FFABS(sample) >= M_PI_2)
189
                    dst[n] = FFSIGN(sample);
190
                else
191
                    dst[n] = sinf(sample);
192
                dst[n] *= gain;
193
            }
194
            break;
195
        case ASC_ERF:
196
            for (int n = 0; n < nb_samples; n++) {
197
                dst[n] = erff(src[n] * factor);
198
                dst[n] *= gain;
199
            }
200
            break;
201
        default:
202
            av_assert0(0);
203
        }
204
    }
205
}
206
207
static void filter_dbl(ASoftClipContext *s,
208
                       void **dptr, const void **sptr,
209
                       int nb_samples, int channels,
210
                       int start, int end)
211
{
212
    double threshold = s->threshold;
213
    double gain = s->output * threshold;
214
    double factor = 1. / threshold;
215
    double param = s->param;
216
217
    for (int c = start; c < end; c++) {
218
        const double *src = sptr[c];
219
        double *dst = dptr[c];
220
221
        switch (s->type) {
222
        case ASC_HARD:
223
            for (int n = 0; n < nb_samples; n++) {
224
                dst[n] = av_clipd(src[n] * factor, -1., 1.);
225
                dst[n] *= gain;
226
            }
227
            break;
228
        case ASC_TANH:
229
            for (int n = 0; n < nb_samples; n++) {
230
                dst[n] = tanh(src[n] * factor * param);
231
                dst[n] *= gain;
232
            }
233
            break;
234
        case ASC_ATAN:
235
            for (int n = 0; n < nb_samples; n++) {
236
                dst[n] = 2. / M_PI * atan(src[n] * factor * param);
237
                dst[n] *= gain;
238
            }
239
            break;
240
        case ASC_CUBIC:
241
            for (int n = 0; n < nb_samples; n++) {
242
                double sample = src[n] * factor;
243
244
                if (FFABS(sample) >= 1.5)
245
                    dst[n] = FFSIGN(sample);
246
                else
247
                    dst[n] = sample - 0.1481 * pow(sample, 3.);
248
                dst[n] *= gain;
249
            }
250
            break;
251
        case ASC_EXP:
252
            for (int n = 0; n < nb_samples; n++) {
253
                dst[n] = 2. / (1. + exp(-2. * src[n] * factor)) - 1.;
254
                dst[n] *= gain;
255
            }
256
            break;
257
        case ASC_ALG:
258
            for (int n = 0; n < nb_samples; n++) {
259
                double sample = src[n] * factor;
260
261
                dst[n] = sample / (sqrt(param + sample * sample));
262
                dst[n] *= gain;
263
            }
264
            break;
265
        case ASC_QUINTIC:
266
            for (int n = 0; n < nb_samples; n++) {
267
                double sample = src[n] * factor;
268
269
                if (FFABS(sample) >= 1.25)
270
                    dst[n] = FFSIGN(sample);
271
                else
272
                    dst[n] = sample - 0.08192 * pow(sample, 5.);
273
                dst[n] *= gain;
274
            }
275
            break;
276
        case ASC_SIN:
277
            for (int n = 0; n < nb_samples; n++) {
278
                double sample = src[n] * factor;
279
280
                if (FFABS(sample) >= M_PI_2)
281
                    dst[n] = FFSIGN(sample);
282
                else
283
                    dst[n] = sin(sample);
284
                dst[n] *= gain;
285
            }
286
            break;
287
        case ASC_ERF:
288
            for (int n = 0; n < nb_samples; n++) {
289
                dst[n] = erf(src[n] * factor);
290
                dst[n] *= gain;
291
            }
292
            break;
293
        default:
294
            av_assert0(0);
295
        }
296
    }
297
}
298
299
static int config_input(AVFilterLink *inlink)
300
{
301
    AVFilterContext *ctx = inlink->dst;
302
    ASoftClipContext *s = ctx->priv;
303
    int ret;
304
305
    switch (inlink->format) {
306
    case AV_SAMPLE_FMT_FLT:
307
    case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
308
    case AV_SAMPLE_FMT_DBL:
309
    case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
310
    default: av_assert0(0);
311
    }
312
313
    if (s->oversample <= 1)
314
        return 0;
315
316
    s->up_ctx = swr_alloc();
317
    s->down_ctx = swr_alloc();
318
    if (!s->up_ctx || !s->down_ctx)
319
        return AVERROR(ENOMEM);
320
321
    av_opt_set_int(s->up_ctx, "in_channel_layout",    inlink->channel_layout, 0);
322
    av_opt_set_int(s->up_ctx, "in_sample_rate",       inlink->sample_rate, 0);
323
    av_opt_set_sample_fmt(s->up_ctx, "in_sample_fmt", inlink->format, 0);
324
325
    av_opt_set_int(s->up_ctx, "out_channel_layout",    inlink->channel_layout, 0);
326
    av_opt_set_int(s->up_ctx, "out_sample_rate",       inlink->sample_rate * s->oversample, 0);
327
    av_opt_set_sample_fmt(s->up_ctx, "out_sample_fmt", inlink->format, 0);
328
329
    av_opt_set_int(s->down_ctx, "in_channel_layout",    inlink->channel_layout, 0);
330
    av_opt_set_int(s->down_ctx, "in_sample_rate",       inlink->sample_rate * s->oversample, 0);
331
    av_opt_set_sample_fmt(s->down_ctx, "in_sample_fmt", inlink->format, 0);
332
333
    av_opt_set_int(s->down_ctx, "out_channel_layout",    inlink->channel_layout, 0);
334
    av_opt_set_int(s->down_ctx, "out_sample_rate",       inlink->sample_rate, 0);
335
    av_opt_set_sample_fmt(s->down_ctx, "out_sample_fmt", inlink->format, 0);
336
337
    ret = swr_init(s->up_ctx);
338
    if (ret < 0)
339
        return ret;
340
341
    ret = swr_init(s->down_ctx);
342
    if (ret < 0)
343
        return ret;
344
345
    return 0;
346
}
347
348
typedef struct ThreadData {
349
    AVFrame *in, *out;
350
    int nb_samples;
351
    int channels;
352
} ThreadData;
353
354
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
355
{
356
    ASoftClipContext *s = ctx->priv;
357
    ThreadData *td = arg;
358
    AVFrame *out = td->out;
359
    AVFrame *in = td->in;
360
    const int channels = td->channels;
361
    const int nb_samples = td->nb_samples;
362
    const int start = (channels * jobnr) / nb_jobs;
363
    const int end = (channels * (jobnr+1)) / nb_jobs;
364
365
    s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
366
              nb_samples, channels, start, end);
367
368
    return 0;
369
}
370
371
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
372
{
373
    AVFilterContext *ctx = inlink->dst;
374
    ASoftClipContext *s = ctx->priv;
375
    AVFilterLink *outlink = ctx->outputs[0];
376
    int ret, nb_samples, channels;
377
    ThreadData td;
378
    AVFrame *out;
379
380
    if (av_frame_is_writable(in)) {
381
        out = in;
382
    } else {
383
        out = ff_get_audio_buffer(outlink, in->nb_samples);
384
        if (!out) {
385
            av_frame_free(&in);
386
            return AVERROR(ENOMEM);
387
        }
388
        av_frame_copy_props(out, in);
389
    }
390
391
    if (av_sample_fmt_is_planar(in->format)) {
392
        nb_samples = in->nb_samples;
393
        channels = in->channels;
394
    } else {
395
        nb_samples = in->channels * in->nb_samples;
396
        channels = 1;
397
    }
398
399
    if (s->oversample > 1) {
400
        s->frame = ff_get_audio_buffer(outlink, in->nb_samples * s->oversample);
401
        if (!s->frame) {
402
            ret = AVERROR(ENOMEM);
403
            goto fail;
404
        }
405
406
        ret = swr_convert(s->up_ctx, (uint8_t**)s->frame->extended_data, in->nb_samples * s->oversample,
407
                          (const uint8_t **)in->extended_data, in->nb_samples);
408
        if (ret < 0)
409
            goto fail;
410
411
        td.in = s->frame;
412
        td.out = s->frame;
413
        td.nb_samples = av_sample_fmt_is_planar(in->format) ? ret : ret * in->channels;
414
        td.channels = channels;
415
        ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(channels,
416
                                                                ff_filter_get_nb_threads(ctx)));
417
418
        ret = swr_convert(s->down_ctx, (uint8_t**)out->extended_data, out->nb_samples,
419
                          (const uint8_t **)s->frame->extended_data, ret);
420
        if (ret < 0)
421
            goto fail;
422
423
        if (out->pts)
424
            out->pts -= s->delay;
425
        s->delay += in->nb_samples - ret;
426
        out->nb_samples = ret;
427
428
        av_frame_free(&s->frame);
429
    } else {
430
        td.in = in;
431
        td.out = out;
432
        td.nb_samples = nb_samples;
433
        td.channels = channels;
434
        ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(channels,
435
                                                                ff_filter_get_nb_threads(ctx)));
436
    }
437
438
    if (out != in)
439
        av_frame_free(&in);
440
441
    return ff_filter_frame(outlink, out);
442
fail:
443
    if (out != in)
444
        av_frame_free(&out);
445
    av_frame_free(&in);
446
    av_frame_free(&s->frame);
447
448
    return ret;
449
}
450
451
static av_cold void uninit(AVFilterContext *ctx)
452
{
453
    ASoftClipContext *s = ctx->priv;
454
455
    swr_free(&s->up_ctx);
456
    swr_free(&s->down_ctx);
457
}
458
459
static const AVFilterPad inputs[] = {
460
    {
461
        .name         = "default",
462
        .type         = AVMEDIA_TYPE_AUDIO,
463
        .filter_frame = filter_frame,
464
        .config_props = config_input,
465
    },
466
    { NULL }
467
};
468
469
static const AVFilterPad outputs[] = {
470
    {
471
        .name = "default",
472
        .type = AVMEDIA_TYPE_AUDIO,
473
    },
474
    { NULL }
475
};
476
477
AVFilter ff_af_asoftclip = {
478
    .name           = "asoftclip",
479
    .description    = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
480
    .query_formats  = query_formats,
481
    .priv_size      = sizeof(ASoftClipContext),
482
    .priv_class     = &asoftclip_class,
483
    .inputs         = inputs,
484
    .outputs        = outputs,
485
    .uninit         = uninit,
486
    .process_command = ff_filter_process_command,
487
    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC |
488
                      AVFILTER_FLAG_SLICE_THREADS,
489
};