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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 
set_expr(AVExpr ** pexpr,const char * expr,void * log_ctx)93 static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
94 {
95     int ret;
96     AVExpr *old = NULL;
97 
98     if (*pexpr)
99         old = *pexpr;
100     ret = av_expr_parse(pexpr, expr, var_names,
101                         NULL, NULL, NULL, NULL, 0, log_ctx);
102     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     av_expr_free(old);
110     return 0;
111 }
112 
init(AVFilterContext * ctx)113 static av_cold int init(AVFilterContext *ctx)
114 {
115     VolumeContext *vol = ctx->priv;
116 
117     vol->fdsp = avpriv_float_dsp_alloc(0);
118     if (!vol->fdsp)
119         return AVERROR(ENOMEM);
120 
121     return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);
122 }
123 
uninit(AVFilterContext * ctx)124 static av_cold void uninit(AVFilterContext *ctx)
125 {
126     VolumeContext *vol = ctx->priv;
127     av_expr_free(vol->volume_pexpr);
128     av_opt_free(vol);
129     av_freep(&vol->fdsp);
130 }
131 
query_formats(AVFilterContext * ctx)132 static int query_formats(AVFilterContext *ctx)
133 {
134     VolumeContext *vol = ctx->priv;
135     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     layouts = ff_all_channel_counts();
161     if (!layouts)
162         return AVERROR(ENOMEM);
163     ret = ff_set_common_channel_layouts(ctx, layouts);
164     if (ret < 0)
165         return ret;
166 
167     formats = ff_make_format_list(sample_fmts[vol->precision]);
168     if (!formats)
169         return AVERROR(ENOMEM);
170     ret = ff_set_common_formats(ctx, formats);
171     if (ret < 0)
172         return ret;
173 
174     formats = ff_all_samplerates();
175     if (!formats)
176         return AVERROR(ENOMEM);
177     return ff_set_common_samplerates(ctx, formats);
178 }
179 
scale_samples_u8(uint8_t * dst,const uint8_t * src,int nb_samples,int volume)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 
scale_samples_u8_small(uint8_t * dst,const uint8_t * src,int nb_samples,int volume)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 
scale_samples_s16(uint8_t * dst,const uint8_t * src,int nb_samples,int volume)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 
scale_samples_s16_small(uint8_t * dst,const uint8_t * src,int nb_samples,int volume)206 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     int16_t *smp_dst       = (int16_t *)dst;
211     const int16_t *smp_src = (const int16_t *)src;
212     for (i = 0; i < nb_samples; i++)
213         smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
214 }
215 
scale_samples_s32(uint8_t * dst,const uint8_t * src,int nb_samples,int volume)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 
volume_init(VolumeContext * vol)226 static av_cold void volume_init(VolumeContext *vol)
227 {
228     vol->samples_align = 1;
229 
230     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     case AV_SAMPLE_FMT_S16:
238         if (vol->volume_i < 0x10000)
239             vol->scale_samples = scale_samples_s16_small;
240         else
241             vol->scale_samples = scale_samples_s16;
242         break;
243     case AV_SAMPLE_FMT_S32:
244         vol->scale_samples = scale_samples_s32;
245         break;
246     case AV_SAMPLE_FMT_FLT:
247         vol->samples_align = 4;
248         break;
249     case AV_SAMPLE_FMT_DBL:
250         vol->samples_align = 8;
251         break;
252     }
253 
254     if (ARCH_X86)
255         ff_volume_init_x86(vol);
256 }
257 
set_volume(AVFilterContext * ctx)258 static int set_volume(AVFilterContext *ctx)
259 {
260     VolumeContext *vol = ctx->priv;
261 
262     vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);
263     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     vol->var_values[VAR_VOLUME] = vol->volume;
273 
274     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            precision_str[vol->precision]);
277 
278     if (vol->precision == PRECISION_FIXED) {
279         vol->volume_i = (int)(vol->volume * 256 + 0.5);
280         vol->volume   = vol->volume_i / 256.0;
281         av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);
282     }
283     av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",
284            vol->volume, 20.0*log10(vol->volume));
285 
286     volume_init(vol);
287     return 0;
288 }
289 
config_output(AVFilterLink * outlink)290 static int config_output(AVFilterLink *outlink)
291 {
292     AVFilterContext *ctx = outlink->src;
293     VolumeContext *vol   = ctx->priv;
294     AVFilterLink *inlink = ctx->inputs[0];
295 
296     vol->sample_fmt = inlink->format;
297     vol->channels   = inlink->channels;
298     vol->planes     = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
299 
300     vol->var_values[VAR_N] =
301     vol->var_values[VAR_NB_CONSUMED_SAMPLES] =
302     vol->var_values[VAR_NB_SAMPLES] =
303     vol->var_values[VAR_POS] =
304     vol->var_values[VAR_PTS] =
305     vol->var_values[VAR_STARTPTS] =
306     vol->var_values[VAR_STARTT] =
307     vol->var_values[VAR_T] =
308     vol->var_values[VAR_VOLUME] = NAN;
309 
310     vol->var_values[VAR_NB_CHANNELS] = inlink->channels;
311     vol->var_values[VAR_TB]          = av_q2d(inlink->time_base);
312     vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;
313 
314     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     return set_volume(ctx);
320 }
321 
process_command(AVFilterContext * ctx,const char * cmd,const char * args,char * res,int res_len,int flags)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 #define D2TS(d)  (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d))
339 #define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))
340 #define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))
341 
filter_frame(AVFilterLink * inlink,AVFrame * buf)342 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
343 {
344     AVFilterContext *ctx = inlink->dst;
345     VolumeContext *vol    = inlink->dst->priv;
346     AVFilterLink *outlink = inlink->dst->outputs[0];
347     int nb_samples        = buf->nb_samples;
348     AVFrame *out_buf;
349     int64_t pos;
350     AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
351     int ret;
352 
353     if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
354         if (vol->replaygain != REPLAYGAIN_DROP) {
355             AVReplayGain *replaygain = (AVReplayGain*)sd->data;
356             int32_t gain  = 100000;
357             uint32_t peak = 100000;
358             float g, p;
359 
360             if (vol->replaygain == REPLAYGAIN_TRACK &&
361                 replaygain->track_gain != INT32_MIN) {
362                 gain = replaygain->track_gain;
363 
364                 if (replaygain->track_peak != 0)
365                     peak = replaygain->track_peak;
366             } else if (replaygain->album_gain != INT32_MIN) {
367                 gain = replaygain->album_gain;
368 
369                 if (replaygain->album_peak != 0)
370                     peak = replaygain->album_peak;
371             } else {
372                 av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
373                        "values are unknown.\n");
374             }
375             g = gain / 100000.0f;
376             p = peak / 100000.0f;
377 
378             av_log(inlink->dst, AV_LOG_VERBOSE,
379                    "Using gain %f dB from replaygain side data.\n", g);
380 
381             vol->volume   = ff_exp10((g + vol->replaygain_preamp) / 20);
382             if (vol->replaygain_noclip)
383                 vol->volume = FFMIN(vol->volume, 1.0 / p);
384             vol->volume_i = (int)(vol->volume * 256 + 0.5);
385 
386             volume_init(vol);
387         }
388         av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
389     }
390 
391     if (isnan(vol->var_values[VAR_STARTPTS])) {
392         vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);
393         vol->var_values[VAR_STARTT  ] = TS2T(buf->pts, inlink->time_base);
394     }
395     vol->var_values[VAR_PTS] = TS2D(buf->pts);
396     vol->var_values[VAR_T  ] = TS2T(buf->pts, inlink->time_base);
397     vol->var_values[VAR_N  ] = inlink->frame_count_out;
398 
399     pos = buf->pkt_pos;
400     vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;
401     if (vol->eval_mode == EVAL_MODE_FRAME)
402         set_volume(ctx);
403 
404     if (vol->volume == 1.0 || vol->volume_i == 256) {
405         out_buf = buf;
406         goto end;
407     }
408 
409     /* do volume scaling in-place if input buffer is writable */
410     if (av_frame_is_writable(buf)
411             && (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {
412         out_buf = buf;
413     } else {
414         out_buf = ff_get_audio_buffer(outlink, nb_samples);
415         if (!out_buf) {
416             av_frame_free(&buf);
417             return AVERROR(ENOMEM);
418         }
419         ret = av_frame_copy_props(out_buf, buf);
420         if (ret < 0) {
421             av_frame_free(&out_buf);
422             av_frame_free(&buf);
423             return ret;
424         }
425     }
426 
427     if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
428         int p, plane_samples;
429 
430         if (av_sample_fmt_is_planar(buf->format))
431             plane_samples = FFALIGN(nb_samples, vol->samples_align);
432         else
433             plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
434 
435         if (vol->precision == PRECISION_FIXED) {
436             for (p = 0; p < vol->planes; p++) {
437                 vol->scale_samples(out_buf->extended_data[p],
438                                    buf->extended_data[p], plane_samples,
439                                    vol->volume_i);
440             }
441         } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
442             for (p = 0; p < vol->planes; p++) {
443                 vol->fdsp->vector_fmul_scalar((float *)out_buf->extended_data[p],
444                                              (const float *)buf->extended_data[p],
445                                              vol->volume, plane_samples);
446             }
447         } else {
448             for (p = 0; p < vol->planes; p++) {
449                 vol->fdsp->vector_dmul_scalar((double *)out_buf->extended_data[p],
450                                              (const double *)buf->extended_data[p],
451                                              vol->volume, plane_samples);
452             }
453         }
454     }
455 
456     emms_c();
457 
458     if (buf != out_buf)
459         av_frame_free(&buf);
460 
461 end:
462     vol->var_values[VAR_NB_CONSUMED_SAMPLES] += out_buf->nb_samples;
463     return ff_filter_frame(outlink, out_buf);
464 }
465 
466 static const AVFilterPad avfilter_af_volume_inputs[] = {
467     {
468         .name           = "default",
469         .type           = AVMEDIA_TYPE_AUDIO,
470         .filter_frame   = filter_frame,
471     },
472     { NULL }
473 };
474 
475 static const AVFilterPad avfilter_af_volume_outputs[] = {
476     {
477         .name         = "default",
478         .type         = AVMEDIA_TYPE_AUDIO,
479         .config_props = config_output,
480     },
481     { NULL }
482 };
483 
484 AVFilter ff_af_volume = {
485     .name           = "volume",
486     .description    = NULL_IF_CONFIG_SMALL("Change input volume."),
487     .query_formats  = query_formats,
488     .priv_size      = sizeof(VolumeContext),
489     .priv_class     = &volume_class,
490     .init           = init,
491     .uninit         = uninit,
492     .inputs         = avfilter_af_volume_inputs,
493     .outputs        = avfilter_af_volume_outputs,
494     .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
495     .process_command = process_command,
496 };
497