FFmpeg  4.3.8
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 
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 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 
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 
125 {
126  VolumeContext *vol = ctx->priv;
128  av_opt_free(vol);
129  av_freep(&vol->fdsp);
130 }
131 
133 {
134  VolumeContext *vol = ctx->priv;
137  static const enum AVSampleFormat sample_fmts[][7] = {
138  [PRECISION_FIXED] = {
146  },
147  [PRECISION_FLOAT] = {
150  AV_SAMPLE_FMT_NONE
151  },
152  [PRECISION_DOUBLE] = {
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 
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 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 
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 
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)
234  else
236  break;
237  case AV_SAMPLE_FMT_S16:
238  if (vol->volume_i < 0x10000)
240  else
242  break;
243  case AV_SAMPLE_FMT_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 
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 
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] =
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],
318 
319  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 #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 
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;
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  }
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 
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  }
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:
463  return ff_filter_frame(outlink, out_buf);
464 }
465 
467  {
468  .name = "default",
469  .type = AVMEDIA_TYPE_AUDIO,
470  .filter_frame = filter_frame,
471  },
472  { NULL }
473 };
474 
476  {
477  .name = "default",
478  .type = AVMEDIA_TYPE_AUDIO,
479  .config_props = config_output,
480  },
481  { NULL }
482 };
483 
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,
496 };
int replaygain
Definition: af_volume.h:77
float, planar
Definition: samplefmt.h:69
#define NULL
Definition: coverity.c:32
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:586
#define A
Definition: af_volume.c:63
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
AVOption.
Definition: opt.h:246
int64_t pkt_pos
reordered pos from the last AVPacket that has been input into the decoder
Definition: frame.h:571
Definition: aeval.c:48
#define AV_LOG_WARNING
Something somehow does not look correct.
Definition: log.h:182
Main libavfilter public API header.
const char * g
Definition: vf_curves.c:115
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_volume.c:124
static const char *const var_names[]
Definition: af_volume.c:46
static av_cold void volume_init(VolumeContext *vol)
Definition: af_volume.c:226
double, planar
Definition: samplefmt.h:70
static av_cold int init(AVFilterContext *ctx)
Definition: af_volume.c:113
static int set_volume(AVFilterContext *ctx)
Definition: af_volume.c:258
int av_expr_parse(AVExpr **expr, const char *s, const char *const *const_names, const char *const *func1_names, double(*const *funcs1)(void *, double), const char *const *func2_names, double(*const *funcs2)(void *, double, double), int log_offset, void *log_ctx)
Parse an expression.
Definition: eval.c:685
double var_values[VAR_VARS_NB]
Definition: af_volume.h:75
#define TS2T(ts, tb)
Definition: af_volume.c:340
uint32_t track_peak
Peak track amplitude, with 100000 representing full scale (but values may overflow).
Definition: replaygain.h:39
AVFilter ff_af_volume
Definition: af_volume.c:484
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:300
AVFrameSideData * av_frame_get_side_data(const AVFrame *frame, enum AVFrameSideDataType type)
Definition: frame.c:739
const char * volume_expr
Definition: af_volume.h:73
#define AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC
Some filters support a generic "enable" expression option that can be used to enable or disable a fil...
Definition: avfilter.h:125
double replaygain_preamp
Definition: af_volume.h:78
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:346
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1075
uint8_t
#define av_cold
Definition: attributes.h:88
AV_SAMPLE_FMT_U8
AVOptions.
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:92
#define emms_c()
Definition: internal.h:55
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:393
Definition: eval.c:157
Structure to hold side data for an AVFrame.
Definition: frame.h:206
int samples_align
Definition: af_volume.h:88
static double av_q2d(AVRational a)
Convert an AVRational to a double.
Definition: rational.h:104
#define AV_LOG_VERBOSE
Detailed information.
Definition: log.h:192
signed 32 bits
Definition: samplefmt.h:62
#define FFALIGN(x, a)
Definition: macros.h:48
int32_t album_gain
Same as track_gain, but for the whole album.
Definition: replaygain.h:43
#define av_log(a,...)
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
static void scale_samples_s32(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:216
A filter pad used for either input or output.
Definition: internal.h:54
#define src
Definition: vp8dsp.c:254
void(* vector_dmul_scalar)(double *dst, const double *src, double mul, int len)
Multiply a vector of double by a scalar double.
Definition: float_dsp.h:100
audio volume filter
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
static av_always_inline double ff_exp10(double x)
Compute 10^x for floating point values.
Definition: ffmath.h:42
#define AV_LOG_ERROR
Something went wrong and cannot losslessly be recovered.
Definition: log.h:176
int ff_set_common_formats(AVFilterContext *ctx, AVFilterFormats *formats)
A helper for query_formats() which sets all links to the same list of formats.
Definition: formats.c:605
av_cold AVFloatDSPContext * avpriv_float_dsp_alloc(int bit_exact)
Allocate a float DSP context.
Definition: float_dsp.c:135
AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)
Request an audio samples buffer with a specific set of permissions.
Definition: audio.c:86
#define ARCH_X86
Definition: config.h:38
#define AVERROR(e)
Definition: error.h:43
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:203
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:188
void * priv
private data for use by the filter
Definition: avfilter.h:353
unsigned int pos
Definition: spdifenc.c:412
AVFILTER_DEFINE_CLASS(volume)
static int query_formats(AVFilterContext *ctx)
Definition: af_volume.c:132
#define F
Definition: af_volume.c:64
static void scale_samples_s16(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:196
audio channel layout utility functions
#define NAN
Definition: mathematics.h:64
#define FFMIN(a, b)
Definition: common.h:96
signed 32 bits, planar
Definition: samplefmt.h:68
static void scale_samples_u8_small(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:188
int replaygain_noclip
Definition: af_volume.h:79
int32_t
AVFormatContext * ctx
Definition: movenc.c:48
Definition: aeval.c:51
unsigned 8 bits, planar
Definition: samplefmt.h:66
static const AVFilterPad avfilter_af_volume_outputs[]
Definition: af_volume.c:475
static void scale_samples_u8(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:180
static const AVFilterPad inputs[]
Definition: af_acontrast.c:193
#define TS2D(ts)
Definition: af_volume.c:339
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
static const char *const precision_str[]
Definition: af_volume.c:42
A list of supported channel layouts.
Definition: formats.h:85
double volume
Definition: af_volume.h:80
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:373
enum AVSampleFormat sample_fmt
Definition: af_volume.h:84
AVFloatDSPContext * fdsp
Definition: af_volume.h:70
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
void av_expr_free(AVExpr *e)
Free a parsed expression previously created with av_expr_parse().
Definition: eval.c:336
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:595
long long int64_t
Definition: coverity.c:34
uint8_t * data
Definition: frame.h:208
void av_frame_remove_side_data(AVFrame *frame, enum AVFrameSideDataType type)
Remove and free all side data instances of the given type.
Definition: frame.c:813
Filter definition.
Definition: avfilter.h:144
#define isnan(x)
Definition: libm.h:340
AVExpr * volume_pexpr
Definition: af_volume.h:74
void(* vector_fmul_scalar)(float *dst, const float *src, float mul, int len)
Multiply a vector of floats by a scalar float.
Definition: float_dsp.h:85
const char * name
Filter name.
Definition: avfilter.h:148
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_volume.c:322
static const AVOption volume_options[]
Definition: af_volume.c:67
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:350
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:439
static void scale_samples_s16_small(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:206
#define flags(name, subs,...)
Definition: cbs_av1.c:576
internal math functions header
int
void av_opt_free(void *obj)
Free all allocated objects in obj.
Definition: opt.c:1610
void ff_volume_init_x86(VolumeContext *vol)
static const AVFilterPad avfilter_af_volume_inputs[]
Definition: af_volume.c:466
common internal and external API header
if(ret< 0)
Definition: vf_mcdeint.c:279
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt)
Get the packed alternative form of the given sample format.
Definition: samplefmt.c:75
signed 16 bits
Definition: samplefmt.h:61
static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
Definition: af_volume.c:342
static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
Definition: af_volume.c:93
uint32_t album_peak
Same as track_peak, but for the whole album,.
Definition: replaygain.h:47
double av_expr_eval(AVExpr *e, const double *const_values, void *opaque)
Evaluate a previously parsed expression.
Definition: eval.c:766
#define OFFSET(x)
Definition: af_volume.c:62
A list of supported formats for one end of a filter link.
Definition: formats.h:64
An instance of a filter.
Definition: avfilter.h:338
#define T
Definition: af_volume.c:65
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:731
#define av_freep(p)
signed 16 bits, planar
Definition: samplefmt.h:67
int32_t track_gain
Track replay gain in microbels (divide by 100000 to get the value in dB).
Definition: replaygain.h:34
formats
Definition: signature.h:48
ReplayGain information in the form of the AVReplayGain struct.
Definition: frame.h:76
internal API functions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:454
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:347
ReplayGain information (see http://wiki.hydrogenaudio.org/index.php?title=ReplayGain_1.0_specification).
Definition: replaygain.h:29
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:593
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:659
void(* scale_samples)(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.h:86
simple arithmetic expression evaluator
static int config_output(AVFilterLink *outlink)
Definition: af_volume.c:290