GCC Code Coverage Report
Directory: ../../../ffmpeg/ Exec Total Coverage
File: src/libavfilter/af_volume.c Lines: 125 216 57.9 %
Date: 2020-08-13 15:06:06 Branches: 47 116 40.5 %

Line Branch Exec Source
1
/*
2
 * Copyright (c) 2011 Stefano Sabatini
3
 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4
 *
5
 * This file is part of FFmpeg.
6
 *
7
 * FFmpeg is free software; you can redistribute it and/or
8
 * modify it under the terms of the GNU Lesser General Public
9
 * License as published by the Free Software Foundation; either
10
 * version 2.1 of the License, or (at your option) any later version.
11
 *
12
 * FFmpeg is distributed in the hope that it will be useful,
13
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
15
 * Lesser General Public License for more details.
16
 *
17
 * You should have received a copy of the GNU Lesser General Public
18
 * License along with FFmpeg; if not, write to the Free Software
19
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20
 */
21
22
/**
23
 * @file
24
 * audio volume filter
25
 */
26
27
#include "libavutil/channel_layout.h"
28
#include "libavutil/common.h"
29
#include "libavutil/eval.h"
30
#include "libavutil/ffmath.h"
31
#include "libavutil/float_dsp.h"
32
#include "libavutil/intreadwrite.h"
33
#include "libavutil/opt.h"
34
#include "libavutil/replaygain.h"
35
36
#include "audio.h"
37
#include "avfilter.h"
38
#include "formats.h"
39
#include "internal.h"
40
#include "af_volume.h"
41
42
static const char * const precision_str[] = {
43
    "fixed", "float", "double"
44
};
45
46
static const char *const var_names[] = {
47
    "n",                   ///< frame number (starting at zero)
48
    "nb_channels",         ///< number of channels
49
    "nb_consumed_samples", ///< number of samples consumed by the filter
50
    "nb_samples",          ///< number of samples in the current frame
51
    "pos",                 ///< position in the file of the frame
52
    "pts",                 ///< frame presentation timestamp
53
    "sample_rate",         ///< sample rate
54
    "startpts",            ///< PTS at start of stream
55
    "startt",              ///< time at start of stream
56
    "t",                   ///< time in the file of the frame
57
    "tb",                  ///< timebase
58
    "volume",              ///< last set value
59
    NULL
60
};
61
62
#define OFFSET(x) offsetof(VolumeContext, x)
63
#define A AV_OPT_FLAG_AUDIO_PARAM
64
#define F AV_OPT_FLAG_FILTERING_PARAM
65
#define T AV_OPT_FLAG_RUNTIME_PARAM
66
67
static const AVOption volume_options[] = {
68
    { "volume", "set volume adjustment expression",
69
            OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F|T },
70
    { "precision", "select mathematical precision",
71
            OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },
72
        { "fixed",  "select 8-bit fixed-point",     0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED  }, INT_MIN, INT_MAX, A|F, "precision" },
73
        { "float",  "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT  }, INT_MIN, INT_MAX, A|F, "precision" },
74
        { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },
75
    { "eval", "specify when to evaluate expressions", OFFSET(eval_mode), AV_OPT_TYPE_INT, {.i64 = EVAL_MODE_ONCE}, 0, EVAL_MODE_NB-1, .flags = A|F, "eval" },
76
         { "once",  "eval volume expression once", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_ONCE},  .flags = A|F, .unit = "eval" },
77
         { "frame", "eval volume expression per-frame",                  0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_FRAME}, .flags = A|F, .unit = "eval" },
78
    { "replaygain", "Apply replaygain side data when present",
79
            OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A|F, "replaygain" },
80
        { "drop",   "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP   }, 0, 0, A|F, "replaygain" },
81
        { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A|F, "replaygain" },
82
        { "track",  "track gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK  }, 0, 0, A|F, "replaygain" },
83
        { "album",  "album gain is preferred",         0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM  }, 0, 0, A|F, "replaygain" },
84
    { "replaygain_preamp", "Apply replaygain pre-amplification",
85
            OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A|F },
86
    { "replaygain_noclip", "Apply replaygain clipping prevention",
87
            OFFSET(replaygain_noclip), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, A|F },
88
    { NULL }
89
};
90
91
AVFILTER_DEFINE_CLASS(volume);
92
93
2
static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
94
{
95
    int ret;
96
2
    AVExpr *old = NULL;
97
98
2
    if (*pexpr)
99
        old = *pexpr;
100
2
    ret = av_expr_parse(pexpr, expr, var_names,
101
                        NULL, NULL, NULL, NULL, 0, log_ctx);
102
2
    if (ret < 0) {
103
        av_log(log_ctx, AV_LOG_ERROR,
104
               "Error when evaluating the volume expression '%s'\n", expr);
105
        *pexpr = old;
106
        return ret;
107
    }
108
109
2
    av_expr_free(old);
110
2
    return 0;
111
}
112
113
2
static av_cold int init(AVFilterContext *ctx)
114
{
115
2
    VolumeContext *vol = ctx->priv;
116
117
2
    vol->fdsp = avpriv_float_dsp_alloc(0);
118
2
    if (!vol->fdsp)
119
        return AVERROR(ENOMEM);
120
121
2
    return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);
122
}
123
124
2
static av_cold void uninit(AVFilterContext *ctx)
125
{
126
2
    VolumeContext *vol = ctx->priv;
127
2
    av_expr_free(vol->volume_pexpr);
128
2
    av_opt_free(vol);
129
2
    av_freep(&vol->fdsp);
130
2
}
131
132
2
static int query_formats(AVFilterContext *ctx)
133
{
134
2
    VolumeContext *vol = ctx->priv;
135
2
    AVFilterFormats *formats = NULL;
136
    AVFilterChannelLayouts *layouts;
137
    static const enum AVSampleFormat sample_fmts[][7] = {
138
        [PRECISION_FIXED] = {
139
            AV_SAMPLE_FMT_U8,
140
            AV_SAMPLE_FMT_U8P,
141
            AV_SAMPLE_FMT_S16,
142
            AV_SAMPLE_FMT_S16P,
143
            AV_SAMPLE_FMT_S32,
144
            AV_SAMPLE_FMT_S32P,
145
            AV_SAMPLE_FMT_NONE
146
        },
147
        [PRECISION_FLOAT] = {
148
            AV_SAMPLE_FMT_FLT,
149
            AV_SAMPLE_FMT_FLTP,
150
            AV_SAMPLE_FMT_NONE
151
        },
152
        [PRECISION_DOUBLE] = {
153
            AV_SAMPLE_FMT_DBL,
154
            AV_SAMPLE_FMT_DBLP,
155
            AV_SAMPLE_FMT_NONE
156
        }
157
    };
158
    int ret;
159
160
2
    layouts = ff_all_channel_counts();
161
2
    if (!layouts)
162
        return AVERROR(ENOMEM);
163
2
    ret = ff_set_common_channel_layouts(ctx, layouts);
164
2
    if (ret < 0)
165
        return ret;
166
167
2
    formats = ff_make_format_list(sample_fmts[vol->precision]);
168
2
    if (!formats)
169
        return AVERROR(ENOMEM);
170
2
    ret = ff_set_common_formats(ctx, formats);
171
2
    if (ret < 0)
172
        return ret;
173
174
2
    formats = ff_all_samplerates();
175
2
    if (!formats)
176
        return AVERROR(ENOMEM);
177
2
    return ff_set_common_samplerates(ctx, formats);
178
}
179
180
static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
181
                                    int nb_samples, int volume)
182
{
183
    int i;
184
    for (i = 0; i < nb_samples; i++)
185
        dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
186
}
187
188
static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
189
                                          int nb_samples, int volume)
190
{
191
    int i;
192
    for (i = 0; i < nb_samples; i++)
193
        dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
194
}
195
196
static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
197
                                     int nb_samples, int volume)
198
{
199
    int i;
200
    int16_t *smp_dst       = (int16_t *)dst;
201
    const int16_t *smp_src = (const int16_t *)src;
202
    for (i = 0; i < nb_samples; i++)
203
        smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
204
}
205
206
259
static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
207
                                           int nb_samples, int volume)
208
{
209
    int i;
210
259
    int16_t *smp_dst       = (int16_t *)dst;
211
259
    const int16_t *smp_src = (const int16_t *)src;
212
529459
    for (i = 0; i < nb_samples; i++)
213
529200
        smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
214
259
}
215
216
static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
217
                                     int nb_samples, int volume)
218
{
219
    int i;
220
    int32_t *smp_dst       = (int32_t *)dst;
221
    const int32_t *smp_src = (const int32_t *)src;
222
    for (i = 0; i < nb_samples; i++)
223
        smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
224
}
225
226
2
static av_cold void volume_init(VolumeContext *vol)
227
{
228
2
    vol->samples_align = 1;
229
230

2
    switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
231
    case AV_SAMPLE_FMT_U8:
232
        if (vol->volume_i < 0x1000000)
233
            vol->scale_samples = scale_samples_u8_small;
234
        else
235
            vol->scale_samples = scale_samples_u8;
236
        break;
237
1
    case AV_SAMPLE_FMT_S16:
238
1
        if (vol->volume_i < 0x10000)
239
1
            vol->scale_samples = scale_samples_s16_small;
240
        else
241
            vol->scale_samples = scale_samples_s16;
242
1
        break;
243
    case AV_SAMPLE_FMT_S32:
244
        vol->scale_samples = scale_samples_s32;
245
        break;
246
1
    case AV_SAMPLE_FMT_FLT:
247
1
        vol->samples_align = 4;
248
1
        break;
249
    case AV_SAMPLE_FMT_DBL:
250
        vol->samples_align = 8;
251
        break;
252
    }
253
254
2
    if (ARCH_X86)
255
2
        ff_volume_init_x86(vol);
256
2
}
257
258
2
static int set_volume(AVFilterContext *ctx)
259
{
260
2
    VolumeContext *vol = ctx->priv;
261
262
2
    vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);
263
2
    if (isnan(vol->volume)) {
264
        if (vol->eval_mode == EVAL_MODE_ONCE) {
265
            av_log(ctx, AV_LOG_ERROR, "Invalid value NaN for volume\n");
266
            return AVERROR(EINVAL);
267
        } else {
268
            av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n");
269
            vol->volume = 0;
270
        }
271
    }
272
2
    vol->var_values[VAR_VOLUME] = vol->volume;
273
274
2
    av_log(ctx, AV_LOG_VERBOSE, "n:%f t:%f pts:%f precision:%s ",
275
           vol->var_values[VAR_N], vol->var_values[VAR_T], vol->var_values[VAR_PTS],
276
2
           precision_str[vol->precision]);
277
278
2
    if (vol->precision == PRECISION_FIXED) {
279
1
        vol->volume_i = (int)(vol->volume * 256 + 0.5);
280
1
        vol->volume   = vol->volume_i / 256.0;
281
1
        av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);
282
    }
283
2
    av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",
284
2
           vol->volume, 20.0*log10(vol->volume));
285
286
2
    volume_init(vol);
287
2
    return 0;
288
}
289
290
2
static int config_output(AVFilterLink *outlink)
291
{
292
2
    AVFilterContext *ctx = outlink->src;
293
2
    VolumeContext *vol   = ctx->priv;
294
2
    AVFilterLink *inlink = ctx->inputs[0];
295
296
2
    vol->sample_fmt = inlink->format;
297
2
    vol->channels   = inlink->channels;
298
2
    vol->planes     = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
299
300
2
    vol->var_values[VAR_N] =
301
2
    vol->var_values[VAR_NB_CONSUMED_SAMPLES] =
302
2
    vol->var_values[VAR_NB_SAMPLES] =
303
2
    vol->var_values[VAR_POS] =
304
2
    vol->var_values[VAR_PTS] =
305
2
    vol->var_values[VAR_STARTPTS] =
306
2
    vol->var_values[VAR_STARTT] =
307
2
    vol->var_values[VAR_T] =
308
2
    vol->var_values[VAR_VOLUME] = NAN;
309
310
2
    vol->var_values[VAR_NB_CHANNELS] = inlink->channels;
311
2
    vol->var_values[VAR_TB]          = av_q2d(inlink->time_base);
312
2
    vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;
313
314
2
    av_log(inlink->src, AV_LOG_VERBOSE, "tb:%f sample_rate:%f nb_channels:%f\n",
315
           vol->var_values[VAR_TB],
316
           vol->var_values[VAR_SAMPLE_RATE],
317
           vol->var_values[VAR_NB_CHANNELS]);
318
319
2
    return set_volume(ctx);
320
}
321
322
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
323
                           char *res, int res_len, int flags)
324
{
325
    VolumeContext *vol = ctx->priv;
326
    int ret = AVERROR(ENOSYS);
327
328
    if (!strcmp(cmd, "volume")) {
329
        if ((ret = set_expr(&vol->volume_pexpr, args, ctx)) < 0)
330
            return ret;
331
        if (vol->eval_mode == EVAL_MODE_ONCE)
332
            set_volume(ctx);
333
    }
334
335
    return ret;
336
}
337
338
300
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
339
{
340
300
    AVFilterContext *ctx = inlink->dst;
341
300
    VolumeContext *vol    = inlink->dst->priv;
342
300
    AVFilterLink *outlink = inlink->dst->outputs[0];
343
300
    int nb_samples        = buf->nb_samples;
344
    AVFrame *out_buf;
345
    int64_t pos;
346
300
    AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
347
    int ret;
348
349

300
    if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
350
        if (vol->replaygain != REPLAYGAIN_DROP) {
351
            AVReplayGain *replaygain = (AVReplayGain*)sd->data;
352
            int32_t gain  = 100000;
353
            uint32_t peak = 100000;
354
            float g, p;
355
356
            if (vol->replaygain == REPLAYGAIN_TRACK &&
357
                replaygain->track_gain != INT32_MIN) {
358
                gain = replaygain->track_gain;
359
360
                if (replaygain->track_peak != 0)
361
                    peak = replaygain->track_peak;
362
            } else if (replaygain->album_gain != INT32_MIN) {
363
                gain = replaygain->album_gain;
364
365
                if (replaygain->album_peak != 0)
366
                    peak = replaygain->album_peak;
367
            } else {
368
                av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
369
                       "values are unknown.\n");
370
            }
371
            g = gain / 100000.0f;
372
            p = peak / 100000.0f;
373
374
            av_log(inlink->dst, AV_LOG_VERBOSE,
375
                   "Using gain %f dB from replaygain side data.\n", g);
376
377
            vol->volume   = ff_exp10((g + vol->replaygain_preamp) / 20);
378
            if (vol->replaygain_noclip)
379
                vol->volume = FFMIN(vol->volume, 1.0 / p);
380
            vol->volume_i = (int)(vol->volume * 256 + 0.5);
381
382
            volume_init(vol);
383
        }
384
        av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
385
    }
386
387
300
    if (isnan(vol->var_values[VAR_STARTPTS])) {
388
2
        vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);
389
2
        vol->var_values[VAR_STARTT  ] = TS2T(buf->pts, inlink->time_base);
390
    }
391
300
    vol->var_values[VAR_PTS] = TS2D(buf->pts);
392
300
    vol->var_values[VAR_T  ] = TS2T(buf->pts, inlink->time_base);
393
300
    vol->var_values[VAR_N  ] = inlink->frame_count_out;
394
395
300
    pos = buf->pkt_pos;
396
300
    vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;
397
300
    if (vol->eval_mode == EVAL_MODE_FRAME)
398
        set_volume(ctx);
399
400

300
    if (vol->volume == 1.0 || vol->volume_i == 256) {
401
        out_buf = buf;
402
        goto end;
403
    }
404
405
    /* do volume scaling in-place if input buffer is writable */
406
300
    if (av_frame_is_writable(buf)
407

300
            && (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {
408
300
        out_buf = buf;
409
    } else {
410
        out_buf = ff_get_audio_buffer(outlink, nb_samples);
411
        if (!out_buf) {
412
            av_frame_free(&buf);
413
            return AVERROR(ENOMEM);
414
        }
415
        ret = av_frame_copy_props(out_buf, buf);
416
        if (ret < 0) {
417
            av_frame_free(&out_buf);
418
            av_frame_free(&buf);
419
            return ret;
420
        }
421
    }
422
423

300
    if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
424
        int p, plane_samples;
425
426
300
        if (av_sample_fmt_is_planar(buf->format))
427
41
            plane_samples = FFALIGN(nb_samples, vol->samples_align);
428
        else
429
259
            plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
430
431
300
        if (vol->precision == PRECISION_FIXED) {
432
518
            for (p = 0; p < vol->planes; p++) {
433
259
                vol->scale_samples(out_buf->extended_data[p],
434
259
                                   buf->extended_data[p], plane_samples,
435
                                   vol->volume_i);
436
            }
437
41
        } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
438
123
            for (p = 0; p < vol->planes; p++) {
439
82
                vol->fdsp->vector_fmul_scalar((float *)out_buf->extended_data[p],
440
82
                                             (const float *)buf->extended_data[p],
441
82
                                             vol->volume, plane_samples);
442
            }
443
        } else {
444
            for (p = 0; p < vol->planes; p++) {
445
                vol->fdsp->vector_dmul_scalar((double *)out_buf->extended_data[p],
446
                                             (const double *)buf->extended_data[p],
447
                                             vol->volume, plane_samples);
448
            }
449
        }
450
    }
451
452
300
    emms_c();
453
454
300
    if (buf != out_buf)
455
        av_frame_free(&buf);
456
457
300
end:
458
300
    vol->var_values[VAR_NB_CONSUMED_SAMPLES] += out_buf->nb_samples;
459
300
    return ff_filter_frame(outlink, out_buf);
460
}
461
462
static const AVFilterPad avfilter_af_volume_inputs[] = {
463
    {
464
        .name           = "default",
465
        .type           = AVMEDIA_TYPE_AUDIO,
466
        .filter_frame   = filter_frame,
467
    },
468
    { NULL }
469
};
470
471
static const AVFilterPad avfilter_af_volume_outputs[] = {
472
    {
473
        .name         = "default",
474
        .type         = AVMEDIA_TYPE_AUDIO,
475
        .config_props = config_output,
476
    },
477
    { NULL }
478
};
479
480
AVFilter ff_af_volume = {
481
    .name           = "volume",
482
    .description    = NULL_IF_CONFIG_SMALL("Change input volume."),
483
    .query_formats  = query_formats,
484
    .priv_size      = sizeof(VolumeContext),
485
    .priv_class     = &volume_class,
486
    .init           = init,
487
    .uninit         = uninit,
488
    .inputs         = avfilter_af_volume_inputs,
489
    .outputs        = avfilter_af_volume_outputs,
490
    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
491
    .process_command = process_command,
492
};