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af_volume.c
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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/float_dsp.h"
31 #include "libavutil/intreadwrite.h"
32 #include "libavutil/opt.h"
33 #include "libavutil/replaygain.h"
34 
35 #include "audio.h"
36 #include "avfilter.h"
37 #include "formats.h"
38 #include "internal.h"
39 #include "af_volume.h"
40 
41 static const char * const precision_str[] = {
42  "fixed", "float", "double"
43 };
44 
45 static const char *const var_names[] = {
46  "n", ///< frame number (starting at zero)
47  "nb_channels", ///< number of channels
48  "nb_consumed_samples", ///< number of samples consumed by the filter
49  "nb_samples", ///< number of samples in the current frame
50  "pos", ///< position in the file of the frame
51  "pts", ///< frame presentation timestamp
52  "sample_rate", ///< sample rate
53  "startpts", ///< PTS at start of stream
54  "startt", ///< time at start of stream
55  "t", ///< time in the file of the frame
56  "tb", ///< timebase
57  "volume", ///< last set value
58  NULL
59 };
60 
61 #define OFFSET(x) offsetof(VolumeContext, x)
62 #define A AV_OPT_FLAG_AUDIO_PARAM
63 #define F AV_OPT_FLAG_FILTERING_PARAM
64 
65 static const AVOption volume_options[] = {
66  { "volume", "set volume adjustment expression",
67  OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F },
68  { "precision", "select mathematical precision",
69  OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },
70  { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" },
71  { "float", "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A|F, "precision" },
72  { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },
73  { "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" },
74  { "once", "eval volume expression once", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_ONCE}, .flags = A|F, .unit = "eval" },
75  { "frame", "eval volume expression per-frame", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_FRAME}, .flags = A|F, .unit = "eval" },
76  { "replaygain", "Apply replaygain side data when present",
77  OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A, "replaygain" },
78  { "drop", "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP }, 0, 0, A, "replaygain" },
79  { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A, "replaygain" },
80  { "track", "track gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK }, 0, 0, A, "replaygain" },
81  { "album", "album gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM }, 0, 0, A, "replaygain" },
82  { "replaygain_preamp", "Apply replaygain pre-amplification",
83  OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A },
84  { "replaygain_noclip", "Apply replaygain clipping prevention",
85  OFFSET(replaygain_noclip), AV_OPT_TYPE_INT, { .i64 = 1 }, 0, 1, A },
86  { NULL },
87 };
88 
89 AVFILTER_DEFINE_CLASS(volume);
90 
91 static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
92 {
93  int ret;
94  AVExpr *old = NULL;
95 
96  if (*pexpr)
97  old = *pexpr;
98  ret = av_expr_parse(pexpr, expr, var_names,
99  NULL, NULL, NULL, NULL, 0, log_ctx);
100  if (ret < 0) {
101  av_log(log_ctx, AV_LOG_ERROR,
102  "Error when evaluating the volume expression '%s'\n", expr);
103  *pexpr = old;
104  return ret;
105  }
106 
107  av_expr_free(old);
108  return 0;
109 }
110 
111 static av_cold int init(AVFilterContext *ctx)
112 {
113  VolumeContext *vol = ctx->priv;
114  return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);
115 }
116 
117 static av_cold void uninit(AVFilterContext *ctx)
118 {
119  VolumeContext *vol = ctx->priv;
120  av_expr_free(vol->volume_pexpr);
121  av_opt_free(vol);
122 }
123 
124 static int query_formats(AVFilterContext *ctx)
125 {
126  VolumeContext *vol = ctx->priv;
127  AVFilterFormats *formats = NULL;
128  AVFilterChannelLayouts *layouts;
129  static const enum AVSampleFormat sample_fmts[][7] = {
130  [PRECISION_FIXED] = {
131  AV_SAMPLE_FMT_U8,
132  AV_SAMPLE_FMT_U8P,
133  AV_SAMPLE_FMT_S16,
134  AV_SAMPLE_FMT_S16P,
135  AV_SAMPLE_FMT_S32,
136  AV_SAMPLE_FMT_S32P,
137  AV_SAMPLE_FMT_NONE
138  },
139  [PRECISION_FLOAT] = {
140  AV_SAMPLE_FMT_FLT,
141  AV_SAMPLE_FMT_FLTP,
142  AV_SAMPLE_FMT_NONE
143  },
144  [PRECISION_DOUBLE] = {
145  AV_SAMPLE_FMT_DBL,
146  AV_SAMPLE_FMT_DBLP,
147  AV_SAMPLE_FMT_NONE
148  }
149  };
150 
151  layouts = ff_all_channel_counts();
152  if (!layouts)
153  return AVERROR(ENOMEM);
154  ff_set_common_channel_layouts(ctx, layouts);
155 
156  formats = ff_make_format_list(sample_fmts[vol->precision]);
157  if (!formats)
158  return AVERROR(ENOMEM);
159  ff_set_common_formats(ctx, formats);
160 
161  formats = ff_all_samplerates();
162  if (!formats)
163  return AVERROR(ENOMEM);
164  ff_set_common_samplerates(ctx, formats);
165 
166  return 0;
167 }
168 
169 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
170  int nb_samples, int volume)
171 {
172  int i;
173  for (i = 0; i < nb_samples; i++)
174  dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
175 }
176 
177 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
178  int nb_samples, int volume)
179 {
180  int i;
181  for (i = 0; i < nb_samples; i++)
182  dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
183 }
184 
185 static inline void scale_samples_s16(uint8_t *dst, const uint8_t *src,
186  int nb_samples, int volume)
187 {
188  int i;
189  int16_t *smp_dst = (int16_t *)dst;
190  const int16_t *smp_src = (const int16_t *)src;
191  for (i = 0; i < nb_samples; i++)
192  smp_dst[i] = av_clip_int16(((int64_t)smp_src[i] * volume + 128) >> 8);
193 }
194 
195 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
196  int nb_samples, int volume)
197 {
198  int i;
199  int16_t *smp_dst = (int16_t *)dst;
200  const int16_t *smp_src = (const int16_t *)src;
201  for (i = 0; i < nb_samples; i++)
202  smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
203 }
204 
205 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
206  int nb_samples, int volume)
207 {
208  int i;
209  int32_t *smp_dst = (int32_t *)dst;
210  const int32_t *smp_src = (const int32_t *)src;
211  for (i = 0; i < nb_samples; i++)
212  smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
213 }
214 
215 static av_cold void volume_init(VolumeContext *vol)
216 {
217  vol->samples_align = 1;
218 
219  switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
220  case AV_SAMPLE_FMT_U8:
221  if (vol->volume_i < 0x1000000)
222  vol->scale_samples = scale_samples_u8_small;
223  else
224  vol->scale_samples = scale_samples_u8;
225  break;
226  case AV_SAMPLE_FMT_S16:
227  if (vol->volume_i < 0x10000)
228  vol->scale_samples = scale_samples_s16_small;
229  else
230  vol->scale_samples = scale_samples_s16;
231  break;
232  case AV_SAMPLE_FMT_S32:
233  vol->scale_samples = scale_samples_s32;
234  break;
235  case AV_SAMPLE_FMT_FLT:
236  avpriv_float_dsp_init(&vol->fdsp, 0);
237  vol->samples_align = 4;
238  break;
239  case AV_SAMPLE_FMT_DBL:
240  avpriv_float_dsp_init(&vol->fdsp, 0);
241  vol->samples_align = 8;
242  break;
243  }
244 
245  if (ARCH_X86)
246  ff_volume_init_x86(vol);
247 }
248 
249 static int set_volume(AVFilterContext *ctx)
250 {
251  VolumeContext *vol = ctx->priv;
252 
253  vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);
254  if (isnan(vol->volume)) {
255  if (vol->eval_mode == EVAL_MODE_ONCE) {
256  av_log(ctx, AV_LOG_ERROR, "Invalid value NaN for volume\n");
257  return AVERROR(EINVAL);
258  } else {
259  av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n");
260  vol->volume = 0;
261  }
262  }
263  vol->var_values[VAR_VOLUME] = vol->volume;
264 
265  av_log(ctx, AV_LOG_VERBOSE, "n:%f t:%f pts:%f precision:%s ",
266  vol->var_values[VAR_N], vol->var_values[VAR_T], vol->var_values[VAR_PTS],
267  precision_str[vol->precision]);
268 
269  if (vol->precision == PRECISION_FIXED) {
270  vol->volume_i = (int)(vol->volume * 256 + 0.5);
271  vol->volume = vol->volume_i / 256.0;
272  av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);
273  }
274  av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",
275  vol->volume, 20.0*log(vol->volume)/M_LN10);
276 
277  volume_init(vol);
278  return 0;
279 }
280 
281 static int config_output(AVFilterLink *outlink)
282 {
283  AVFilterContext *ctx = outlink->src;
284  VolumeContext *vol = ctx->priv;
285  AVFilterLink *inlink = ctx->inputs[0];
286 
287  vol->sample_fmt = inlink->format;
288  vol->channels = inlink->channels;
289  vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
290 
291  vol->var_values[VAR_N] =
292  vol->var_values[VAR_NB_CONSUMED_SAMPLES] =
293  vol->var_values[VAR_NB_SAMPLES] =
294  vol->var_values[VAR_POS] =
295  vol->var_values[VAR_PTS] =
296  vol->var_values[VAR_STARTPTS] =
297  vol->var_values[VAR_STARTT] =
298  vol->var_values[VAR_T] =
299  vol->var_values[VAR_VOLUME] = NAN;
300 
301  vol->var_values[VAR_NB_CHANNELS] = inlink->channels;
302  vol->var_values[VAR_TB] = av_q2d(inlink->time_base);
303  vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;
304 
305  av_log(inlink->src, AV_LOG_VERBOSE, "tb:%f sample_rate:%f nb_channels:%f\n",
306  vol->var_values[VAR_TB],
307  vol->var_values[VAR_SAMPLE_RATE],
308  vol->var_values[VAR_NB_CHANNELS]);
309 
310  return set_volume(ctx);
311 }
312 
313 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
314  char *res, int res_len, int flags)
315 {
316  VolumeContext *vol = ctx->priv;
317  int ret = AVERROR(ENOSYS);
318 
319  if (!strcmp(cmd, "volume")) {
320  if ((ret = set_expr(&vol->volume_pexpr, args, ctx)) < 0)
321  return ret;
322  if (vol->eval_mode == EVAL_MODE_ONCE)
323  set_volume(ctx);
324  }
325 
326  return ret;
327 }
328 
329 #define D2TS(d) (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d))
330 #define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))
331 #define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))
332 
333 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
334 {
335  AVFilterContext *ctx = inlink->dst;
336  VolumeContext *vol = inlink->dst->priv;
337  AVFilterLink *outlink = inlink->dst->outputs[0];
338  int nb_samples = buf->nb_samples;
339  AVFrame *out_buf;
340  int64_t pos;
341  AVFrameSideData *sd = av_frame_get_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
342  int ret;
343 
344  if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
345  if (vol->replaygain != REPLAYGAIN_DROP) {
346  AVReplayGain *replaygain = (AVReplayGain*)sd->data;
347  int32_t gain = 100000;
348  uint32_t peak = 100000;
349  float g, p;
350 
351  if (vol->replaygain == REPLAYGAIN_TRACK &&
352  replaygain->track_gain != INT32_MIN) {
353  gain = replaygain->track_gain;
354 
355  if (replaygain->track_peak != 0)
356  peak = replaygain->track_peak;
357  } else if (replaygain->album_gain != INT32_MIN) {
358  gain = replaygain->album_gain;
359 
360  if (replaygain->album_peak != 0)
361  peak = replaygain->album_peak;
362  } else {
363  av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
364  "values are unknown.\n");
365  }
366  g = gain / 100000.0f;
367  p = peak / 100000.0f;
368 
369  av_log(inlink->dst, AV_LOG_VERBOSE,
370  "Using gain %f dB from replaygain side data.\n", g);
371 
372  vol->volume = pow(10, (g + vol->replaygain_preamp) / 20);
373  if (vol->replaygain_noclip)
374  vol->volume = FFMIN(vol->volume, 1.0 / p);
375  vol->volume_i = (int)(vol->volume * 256 + 0.5);
376 
377  volume_init(vol);
378  }
379  av_frame_remove_side_data(buf, AV_FRAME_DATA_REPLAYGAIN);
380  }
381 
382  if (isnan(vol->var_values[VAR_STARTPTS])) {
383  vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);
384  vol->var_values[VAR_STARTT ] = TS2T(buf->pts, inlink->time_base);
385  }
386  vol->var_values[VAR_PTS] = TS2D(buf->pts);
387  vol->var_values[VAR_T ] = TS2T(buf->pts, inlink->time_base);
388  vol->var_values[VAR_N ] = inlink->frame_count;
389 
390  pos = av_frame_get_pkt_pos(buf);
391  vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;
392  if (vol->eval_mode == EVAL_MODE_FRAME)
393  set_volume(ctx);
394 
395  if (vol->volume == 1.0 || vol->volume_i == 256) {
396  out_buf = buf;
397  goto end;
398  }
399 
400  /* do volume scaling in-place if input buffer is writable */
401  if (av_frame_is_writable(buf)) {
402  out_buf = buf;
403  } else {
404  out_buf = ff_get_audio_buffer(inlink, nb_samples);
405  if (!out_buf)
406  return AVERROR(ENOMEM);
407  ret = av_frame_copy_props(out_buf, buf);
408  if (ret < 0) {
409  av_frame_free(&out_buf);
410  av_frame_free(&buf);
411  return ret;
412  }
413  }
414 
415  if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
416  int p, plane_samples;
417 
418  if (av_sample_fmt_is_planar(buf->format))
419  plane_samples = FFALIGN(nb_samples, vol->samples_align);
420  else
421  plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
422 
423  if (vol->precision == PRECISION_FIXED) {
424  for (p = 0; p < vol->planes; p++) {
425  vol->scale_samples(out_buf->extended_data[p],
426  buf->extended_data[p], plane_samples,
427  vol->volume_i);
428  }
429  } else if (av_get_packed_sample_fmt(vol->sample_fmt) == AV_SAMPLE_FMT_FLT) {
430  for (p = 0; p < vol->planes; p++) {
431  vol->fdsp.vector_fmul_scalar((float *)out_buf->extended_data[p],
432  (const float *)buf->extended_data[p],
433  vol->volume, plane_samples);
434  }
435  } else {
436  for (p = 0; p < vol->planes; p++) {
437  vol->fdsp.vector_dmul_scalar((double *)out_buf->extended_data[p],
438  (const double *)buf->extended_data[p],
439  vol->volume, plane_samples);
440  }
441  }
442  }
443 
444  emms_c();
445 
446  if (buf != out_buf)
447  av_frame_free(&buf);
448 
449 end:
450  vol->var_values[VAR_NB_CONSUMED_SAMPLES] += out_buf->nb_samples;
451  return ff_filter_frame(outlink, out_buf);
452 }
453 
454 static const AVFilterPad avfilter_af_volume_inputs[] = {
455  {
456  .name = "default",
457  .type = AVMEDIA_TYPE_AUDIO,
458  .filter_frame = filter_frame,
459  },
460  { NULL }
461 };
462 
463 static const AVFilterPad avfilter_af_volume_outputs[] = {
464  {
465  .name = "default",
466  .type = AVMEDIA_TYPE_AUDIO,
467  .config_props = config_output,
468  },
469  { NULL }
470 };
471 
472 AVFilter ff_af_volume = {
473  .name = "volume",
474  .description = NULL_IF_CONFIG_SMALL("Change input volume."),
475  .query_formats = query_formats,
476  .priv_size = sizeof(VolumeContext),
477  .priv_class = &volume_class,
478  .init = init,
479  .uninit = uninit,
480  .inputs = avfilter_af_volume_inputs,
481  .outputs = avfilter_af_volume_outputs,
482  .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
483  .process_command = process_command,
484 };
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