FFmpeg
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 
66 static const AVOption volume_options[] = {
67  { "volume", "set volume adjustment expression",
68  OFFSET(volume_expr), AV_OPT_TYPE_STRING, { .str = "1.0" }, .flags = A|F },
69  { "precision", "select mathematical precision",
70  OFFSET(precision), AV_OPT_TYPE_INT, { .i64 = PRECISION_FLOAT }, PRECISION_FIXED, PRECISION_DOUBLE, A|F, "precision" },
71  { "fixed", "select 8-bit fixed-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FIXED }, INT_MIN, INT_MAX, A|F, "precision" },
72  { "float", "select 32-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_FLOAT }, INT_MIN, INT_MAX, A|F, "precision" },
73  { "double", "select 64-bit floating-point", 0, AV_OPT_TYPE_CONST, { .i64 = PRECISION_DOUBLE }, INT_MIN, INT_MAX, A|F, "precision" },
74  { "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" },
75  { "once", "eval volume expression once", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_ONCE}, .flags = A|F, .unit = "eval" },
76  { "frame", "eval volume expression per-frame", 0, AV_OPT_TYPE_CONST, {.i64=EVAL_MODE_FRAME}, .flags = A|F, .unit = "eval" },
77  { "replaygain", "Apply replaygain side data when present",
78  OFFSET(replaygain), AV_OPT_TYPE_INT, { .i64 = REPLAYGAIN_DROP }, REPLAYGAIN_DROP, REPLAYGAIN_ALBUM, A|F, "replaygain" },
79  { "drop", "replaygain side data is dropped", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_DROP }, 0, 0, A|F, "replaygain" },
80  { "ignore", "replaygain side data is ignored", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_IGNORE }, 0, 0, A|F, "replaygain" },
81  { "track", "track gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_TRACK }, 0, 0, A|F, "replaygain" },
82  { "album", "album gain is preferred", 0, AV_OPT_TYPE_CONST, { .i64 = REPLAYGAIN_ALBUM }, 0, 0, A|F, "replaygain" },
83  { "replaygain_preamp", "Apply replaygain pre-amplification",
84  OFFSET(replaygain_preamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.0 }, -15.0, 15.0, A|F },
85  { "replaygain_noclip", "Apply replaygain clipping prevention",
86  OFFSET(replaygain_noclip), AV_OPT_TYPE_BOOL, { .i64 = 1 }, 0, 1, A|F },
87  { NULL }
88 };
89 
90 AVFILTER_DEFINE_CLASS(volume);
91 
92 static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
93 {
94  int ret;
95  AVExpr *old = NULL;
96 
97  if (*pexpr)
98  old = *pexpr;
99  ret = av_expr_parse(pexpr, expr, var_names,
100  NULL, NULL, NULL, NULL, 0, log_ctx);
101  if (ret < 0) {
102  av_log(log_ctx, AV_LOG_ERROR,
103  "Error when evaluating the volume expression '%s'\n", expr);
104  *pexpr = old;
105  return ret;
106  }
107 
108  av_expr_free(old);
109  return 0;
110 }
111 
113 {
114  VolumeContext *vol = ctx->priv;
115 
116  vol->fdsp = avpriv_float_dsp_alloc(0);
117  if (!vol->fdsp)
118  return AVERROR(ENOMEM);
119 
120  return set_expr(&vol->volume_pexpr, vol->volume_expr, ctx);
121 }
122 
124 {
125  VolumeContext *vol = ctx->priv;
127  av_opt_free(vol);
128  av_freep(&vol->fdsp);
129 }
130 
132 {
133  VolumeContext *vol = ctx->priv;
136  static const enum AVSampleFormat sample_fmts[][7] = {
137  [PRECISION_FIXED] = {
145  },
146  [PRECISION_FLOAT] = {
149  AV_SAMPLE_FMT_NONE
150  },
151  [PRECISION_DOUBLE] = {
154  AV_SAMPLE_FMT_NONE
155  }
156  };
157  int ret;
158 
159  layouts = ff_all_channel_counts();
160  if (!layouts)
161  return AVERROR(ENOMEM);
162  ret = ff_set_common_channel_layouts(ctx, layouts);
163  if (ret < 0)
164  return ret;
165 
166  formats = ff_make_format_list(sample_fmts[vol->precision]);
167  if (!formats)
168  return AVERROR(ENOMEM);
169  ret = ff_set_common_formats(ctx, formats);
170  if (ret < 0)
171  return ret;
172 
173  formats = ff_all_samplerates();
174  if (!formats)
175  return AVERROR(ENOMEM);
176  return ff_set_common_samplerates(ctx, formats);
177 }
178 
179 static inline void scale_samples_u8(uint8_t *dst, const uint8_t *src,
180  int nb_samples, int volume)
181 {
182  int i;
183  for (i = 0; i < nb_samples; i++)
184  dst[i] = av_clip_uint8(((((int64_t)src[i] - 128) * volume + 128) >> 8) + 128);
185 }
186 
187 static inline void scale_samples_u8_small(uint8_t *dst, const uint8_t *src,
188  int nb_samples, int volume)
189 {
190  int i;
191  for (i = 0; i < nb_samples; i++)
192  dst[i] = av_clip_uint8((((src[i] - 128) * volume + 128) >> 8) + 128);
193 }
194 
195 static inline void scale_samples_s16(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(((int64_t)smp_src[i] * volume + 128) >> 8);
203 }
204 
205 static inline void scale_samples_s16_small(uint8_t *dst, const uint8_t *src,
206  int nb_samples, int volume)
207 {
208  int i;
209  int16_t *smp_dst = (int16_t *)dst;
210  const int16_t *smp_src = (const int16_t *)src;
211  for (i = 0; i < nb_samples; i++)
212  smp_dst[i] = av_clip_int16((smp_src[i] * volume + 128) >> 8);
213 }
214 
215 static inline void scale_samples_s32(uint8_t *dst, const uint8_t *src,
216  int nb_samples, int volume)
217 {
218  int i;
219  int32_t *smp_dst = (int32_t *)dst;
220  const int32_t *smp_src = (const int32_t *)src;
221  for (i = 0; i < nb_samples; i++)
222  smp_dst[i] = av_clipl_int32((((int64_t)smp_src[i] * volume + 128) >> 8));
223 }
224 
226 {
227  vol->samples_align = 1;
228 
229  switch (av_get_packed_sample_fmt(vol->sample_fmt)) {
230  case AV_SAMPLE_FMT_U8:
231  if (vol->volume_i < 0x1000000)
233  else
235  break;
236  case AV_SAMPLE_FMT_S16:
237  if (vol->volume_i < 0x10000)
239  else
241  break;
242  case AV_SAMPLE_FMT_S32:
244  break;
245  case AV_SAMPLE_FMT_FLT:
246  vol->samples_align = 4;
247  break;
248  case AV_SAMPLE_FMT_DBL:
249  vol->samples_align = 8;
250  break;
251  }
252 
253  if (ARCH_X86)
254  ff_volume_init_x86(vol);
255 }
256 
258 {
259  VolumeContext *vol = ctx->priv;
260 
261  vol->volume = av_expr_eval(vol->volume_pexpr, vol->var_values, NULL);
262  if (isnan(vol->volume)) {
263  if (vol->eval_mode == EVAL_MODE_ONCE) {
264  av_log(ctx, AV_LOG_ERROR, "Invalid value NaN for volume\n");
265  return AVERROR(EINVAL);
266  } else {
267  av_log(ctx, AV_LOG_WARNING, "Invalid value NaN for volume, setting to 0\n");
268  vol->volume = 0;
269  }
270  }
271  vol->var_values[VAR_VOLUME] = vol->volume;
272 
273  av_log(ctx, AV_LOG_VERBOSE, "n:%f t:%f pts:%f precision:%s ",
274  vol->var_values[VAR_N], vol->var_values[VAR_T], vol->var_values[VAR_PTS],
275  precision_str[vol->precision]);
276 
277  if (vol->precision == PRECISION_FIXED) {
278  vol->volume_i = (int)(vol->volume * 256 + 0.5);
279  vol->volume = vol->volume_i / 256.0;
280  av_log(ctx, AV_LOG_VERBOSE, "volume_i:%d/255 ", vol->volume_i);
281  }
282  av_log(ctx, AV_LOG_VERBOSE, "volume:%f volume_dB:%f\n",
283  vol->volume, 20.0*log10(vol->volume));
284 
285  volume_init(vol);
286  return 0;
287 }
288 
289 static int config_output(AVFilterLink *outlink)
290 {
291  AVFilterContext *ctx = outlink->src;
292  VolumeContext *vol = ctx->priv;
293  AVFilterLink *inlink = ctx->inputs[0];
294 
295  vol->sample_fmt = inlink->format;
296  vol->channels = inlink->channels;
297  vol->planes = av_sample_fmt_is_planar(inlink->format) ? vol->channels : 1;
298 
299  vol->var_values[VAR_N] =
301  vol->var_values[VAR_NB_SAMPLES] =
302  vol->var_values[VAR_POS] =
303  vol->var_values[VAR_PTS] =
304  vol->var_values[VAR_STARTPTS] =
305  vol->var_values[VAR_STARTT] =
306  vol->var_values[VAR_T] =
307  vol->var_values[VAR_VOLUME] = NAN;
308 
309  vol->var_values[VAR_NB_CHANNELS] = inlink->channels;
310  vol->var_values[VAR_TB] = av_q2d(inlink->time_base);
311  vol->var_values[VAR_SAMPLE_RATE] = inlink->sample_rate;
312 
313  av_log(inlink->src, AV_LOG_VERBOSE, "tb:%f sample_rate:%f nb_channels:%f\n",
314  vol->var_values[VAR_TB],
317 
318  return set_volume(ctx);
319 }
320 
321 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
322  char *res, int res_len, int flags)
323 {
324  VolumeContext *vol = ctx->priv;
325  int ret = AVERROR(ENOSYS);
326 
327  if (!strcmp(cmd, "volume")) {
328  if ((ret = set_expr(&vol->volume_pexpr, args, ctx)) < 0)
329  return ret;
330  if (vol->eval_mode == EVAL_MODE_ONCE)
331  set_volume(ctx);
332  }
333 
334  return ret;
335 }
336 
337 #define D2TS(d) (isnan(d) ? AV_NOPTS_VALUE : (int64_t)(d))
338 #define TS2D(ts) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts))
339 #define TS2T(ts, tb) ((ts) == AV_NOPTS_VALUE ? NAN : (double)(ts)*av_q2d(tb))
340 
342 {
343  AVFilterContext *ctx = inlink->dst;
344  VolumeContext *vol = inlink->dst->priv;
345  AVFilterLink *outlink = inlink->dst->outputs[0];
346  int nb_samples = buf->nb_samples;
347  AVFrame *out_buf;
348  int64_t pos;
350  int ret;
351 
352  if (sd && vol->replaygain != REPLAYGAIN_IGNORE) {
353  if (vol->replaygain != REPLAYGAIN_DROP) {
354  AVReplayGain *replaygain = (AVReplayGain*)sd->data;
355  int32_t gain = 100000;
356  uint32_t peak = 100000;
357  float g, p;
358 
359  if (vol->replaygain == REPLAYGAIN_TRACK &&
360  replaygain->track_gain != INT32_MIN) {
361  gain = replaygain->track_gain;
362 
363  if (replaygain->track_peak != 0)
364  peak = replaygain->track_peak;
365  } else if (replaygain->album_gain != INT32_MIN) {
366  gain = replaygain->album_gain;
367 
368  if (replaygain->album_peak != 0)
369  peak = replaygain->album_peak;
370  } else {
371  av_log(inlink->dst, AV_LOG_WARNING, "Both ReplayGain gain "
372  "values are unknown.\n");
373  }
374  g = gain / 100000.0f;
375  p = peak / 100000.0f;
376 
377  av_log(inlink->dst, AV_LOG_VERBOSE,
378  "Using gain %f dB from replaygain side data.\n", g);
379 
380  vol->volume = ff_exp10((g + vol->replaygain_preamp) / 20);
381  if (vol->replaygain_noclip)
382  vol->volume = FFMIN(vol->volume, 1.0 / p);
383  vol->volume_i = (int)(vol->volume * 256 + 0.5);
384 
385  volume_init(vol);
386  }
388  }
389 
390  if (isnan(vol->var_values[VAR_STARTPTS])) {
391  vol->var_values[VAR_STARTPTS] = TS2D(buf->pts);
392  vol->var_values[VAR_STARTT ] = TS2T(buf->pts, inlink->time_base);
393  }
394  vol->var_values[VAR_PTS] = TS2D(buf->pts);
395  vol->var_values[VAR_T ] = TS2T(buf->pts, inlink->time_base);
396  vol->var_values[VAR_N ] = inlink->frame_count_out;
397 
398  pos = buf->pkt_pos;
399  vol->var_values[VAR_POS] = pos == -1 ? NAN : pos;
400  if (vol->eval_mode == EVAL_MODE_FRAME)
401  set_volume(ctx);
402 
403  if (vol->volume == 1.0 || vol->volume_i == 256) {
404  out_buf = buf;
405  goto end;
406  }
407 
408  /* do volume scaling in-place if input buffer is writable */
409  if (av_frame_is_writable(buf)
410  && (vol->precision != PRECISION_FIXED || vol->volume_i > 0)) {
411  out_buf = buf;
412  } else {
413  out_buf = ff_get_audio_buffer(outlink, nb_samples);
414  if (!out_buf) {
415  av_frame_free(&buf);
416  return AVERROR(ENOMEM);
417  }
418  ret = av_frame_copy_props(out_buf, buf);
419  if (ret < 0) {
420  av_frame_free(&out_buf);
421  av_frame_free(&buf);
422  return ret;
423  }
424  }
425 
426  if (vol->precision != PRECISION_FIXED || vol->volume_i > 0) {
427  int p, plane_samples;
428 
430  plane_samples = FFALIGN(nb_samples, vol->samples_align);
431  else
432  plane_samples = FFALIGN(nb_samples * vol->channels, vol->samples_align);
433 
434  if (vol->precision == PRECISION_FIXED) {
435  for (p = 0; p < vol->planes; p++) {
436  vol->scale_samples(out_buf->extended_data[p],
437  buf->extended_data[p], plane_samples,
438  vol->volume_i);
439  }
441  for (p = 0; p < vol->planes; p++) {
442  vol->fdsp->vector_fmul_scalar((float *)out_buf->extended_data[p],
443  (const float *)buf->extended_data[p],
444  vol->volume, plane_samples);
445  }
446  } else {
447  for (p = 0; p < vol->planes; p++) {
448  vol->fdsp->vector_dmul_scalar((double *)out_buf->extended_data[p],
449  (const double *)buf->extended_data[p],
450  vol->volume, plane_samples);
451  }
452  }
453  }
454 
455  emms_c();
456 
457  if (buf != out_buf)
458  av_frame_free(&buf);
459 
460 end:
462  return ff_filter_frame(outlink, out_buf);
463 }
464 
466  {
467  .name = "default",
468  .type = AVMEDIA_TYPE_AUDIO,
469  .filter_frame = filter_frame,
470  },
471  { NULL }
472 };
473 
475  {
476  .name = "default",
477  .type = AVMEDIA_TYPE_AUDIO,
478  .config_props = config_output,
479  },
480  { NULL }
481 };
482 
484  .name = "volume",
485  .description = NULL_IF_CONFIG_SMALL("Change input volume."),
486  .query_formats = query_formats,
487  .priv_size = sizeof(VolumeContext),
488  .priv_class = &volume_class,
489  .init = init,
490  .uninit = uninit,
491  .inputs = avfilter_af_volume_inputs,
492  .outputs = avfilter_af_volume_outputs,
495 };
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:549
#define A
Definition: af_volume.c:63
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
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:566
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:123
static const char *const var_names[]
Definition: af_volume.c:46
static av_cold void volume_init(VolumeContext *vol)
Definition: af_volume.c:225
double, planar
Definition: samplefmt.h:70
static av_cold int init(AVFilterContext *ctx)
Definition: af_volume.c:112
static int set_volume(AVFilterContext *ctx)
Definition: af_volume.c:257
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:679
double var_values[VAR_VARS_NB]
Definition: af_volume.h:75
#define src
Definition: vp8dsp.c:254
#define TS2T(ts, tb)
Definition: af_volume.c:339
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:483
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
AVFrameSideData * av_frame_get_side_data(const AVFrame *frame, enum AVFrameSideDataType type)
Definition: frame.c:734
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:1080
uint8_t
#define av_cold
Definition: attributes.h:82
AV_SAMPLE_FMT_U8
AVOptions.
static av_cold int end(AVCodecContext *avctx)
Definition: avrndec.c:90
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
Definition: eval.c:157
Structure to hold side data for an AVFrame.
Definition: frame.h:201
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:215
A filter pad used for either input or output.
Definition: internal.h:54
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:259
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:568
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
void av_frame_free(AVFrame **frame)
Free the frame and any dynamically allocated objects in it, e.g.
Definition: frame.c:202
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
void * priv
private data for use by the filter
Definition: avfilter.h:353
AVFILTER_DEFINE_CLASS(volume)
static int query_formats(AVFilterContext *ctx)
Definition: af_volume.c:131
#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:195
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:187
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:474
static void scale_samples_u8(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:179
#define TS2D(ts)
Definition: af_volume.c:338
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
if(ret)
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:368
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:334
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
uint8_t * data
Definition: frame.h:203
void av_frame_remove_side_data(AVFrame *frame, enum AVFrameSideDataType type)
If side data of the supplied type exists in the frame, free it and remove it from the frame...
Definition: frame.c:805
void * buf
Definition: avisynth_c.h:766
these buffered frames must be flushed immediately if a new input produces new the filter must not call request_frame to get more It must just process the frame or queue it The task of requesting more frames is left to the filter s request_frame method or the application If a filter has several inputs
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:321
static const AVOption volume_options[]
Definition: af_volume.c:66
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:395
static void scale_samples_s16_small(uint8_t *dst, const uint8_t *src, int nb_samples, int volume)
Definition: af_volume.c:205
#define flags(name, subs,...)
Definition: cbs_av1.c:561
The exact code depends on how similar the blocks are and how related they are to the and needs to apply these operations to the correct inlink or outlink if there are several Macros are available to factor that when no extra processing is inlink
internal math functions header
int
void av_opt_free(void *obj)
Free all allocated objects in obj.
Definition: opt.c:1558
void ff_volume_init_x86(VolumeContext *vol)
static const AVFilterPad avfilter_af_volume_inputs[]
Definition: af_volume.c:465
common internal and external API header
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:341
static int set_expr(AVExpr **pexpr, const char *expr, void *log_ctx)
Definition: af_volume.c:92
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:734
#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
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
#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
Filter the word “frame” indicates either a video frame or a group of audio as stored in an AVFrame structure Format for each input and each output the list of supported formats For video that means pixel format For audio that means channel sample they are references to shared objects When the negotiation mechanism computes the intersection of the formats supported at each end of a all references to both lists are replaced with a reference to the intersection And when a single format is eventually chosen for a link amongst the remaining all references to the list are updated That means that if a filter requires that its input and output have the same format amongst a supported all it has to do is use a reference to the same list of formats query_formats can leave some formats unset and return AVERROR(EAGAIN) to cause the negotiation mechanism toagain later.That can be used by filters with complex requirements to use the format negotiated on one link to set the formats supported on another.Frame references ownership and permissions
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
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:361
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:654
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:289