FFmpeg  4.2.3
af_asoftclip.c
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1 /*
2  * Copyright (c) 2019 The FFmpeg Project
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public
8  * License as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14  * Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public
17  * License along with FFmpeg; if not, write to the Free Software
18  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
22 #include "libavutil/opt.h"
23 #include "avfilter.h"
24 #include "audio.h"
25 #include "formats.h"
26 
36 };
37 
38 typedef struct ASoftClipContext {
39  const AVClass *class;
40 
41  int type;
42  double param;
43 
44  void (*filter)(struct ASoftClipContext *s, void **dst, const void **src,
45  int nb_samples, int channels);
47 
48 #define OFFSET(x) offsetof(ASoftClipContext, x)
49 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
50 
51 static const AVOption asoftclip_options[] = {
52  { "type", "set softclip type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=0}, 0, NB_TYPES-1, A, "types" },
53  { "tanh", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_TANH}, 0, 0, A, "types" },
54  { "atan", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ATAN}, 0, 0, A, "types" },
55  { "cubic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_CUBIC}, 0, 0, A, "types" },
56  { "exp", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_EXP}, 0, 0, A, "types" },
57  { "alg", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_ALG}, 0, 0, A, "types" },
58  { "quintic", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_QUINTIC},0, 0, A, "types" },
59  { "sin", NULL, 0, AV_OPT_TYPE_CONST, {.i64=ASC_SIN}, 0, 0, A, "types" },
60  { "param", "set softclip parameter", OFFSET(param), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.01, 3, A },
61  { NULL }
62 };
63 
64 AVFILTER_DEFINE_CLASS(asoftclip);
65 
67 {
70  static const enum AVSampleFormat sample_fmts[] = {
74  };
75  int ret;
76 
77  formats = ff_make_format_list(sample_fmts);
78  if (!formats)
79  return AVERROR(ENOMEM);
80  ret = ff_set_common_formats(ctx, formats);
81  if (ret < 0)
82  return ret;
83 
84  layouts = ff_all_channel_counts();
85  if (!layouts)
86  return AVERROR(ENOMEM);
87 
88  ret = ff_set_common_channel_layouts(ctx, layouts);
89  if (ret < 0)
90  return ret;
91 
92  formats = ff_all_samplerates();
93  return ff_set_common_samplerates(ctx, formats);
94 }
95 
96 #define SQR(x) ((x) * (x))
97 
99  void **dptr, const void **sptr,
100  int nb_samples, int channels)
101 {
102  float param = s->param;
103 
104  for (int c = 0; c < channels; c++) {
105  const float *src = sptr[c];
106  float *dst = dptr[c];
107 
108  switch (s->type) {
109  case ASC_TANH:
110  for (int n = 0; n < nb_samples; n++) {
111  dst[n] = tanhf(src[n] * param);
112  }
113  break;
114  case ASC_ATAN:
115  for (int n = 0; n < nb_samples; n++)
116  dst[n] = 2.f / M_PI * atanf(src[n] * param);
117  break;
118  case ASC_CUBIC:
119  for (int n = 0; n < nb_samples; n++) {
120  if (FFABS(src[n]) >= 1.5f)
121  dst[n] = FFSIGN(src[n]);
122  else
123  dst[n] = src[n] - 0.1481f * powf(src[n], 3.f);
124  }
125  break;
126  case ASC_EXP:
127  for (int n = 0; n < nb_samples; n++)
128  dst[n] = 2.f / (1.f + expf(-2.f * src[n])) - 1.;
129  break;
130  case ASC_ALG:
131  for (int n = 0; n < nb_samples; n++)
132  dst[n] = src[n] / (sqrtf(param + src[n] * src[n]));
133  break;
134  case ASC_QUINTIC:
135  for (int n = 0; n < nb_samples; n++) {
136  if (FFABS(src[n]) >= 1.25)
137  dst[n] = FFSIGN(src[n]);
138  else
139  dst[n] = src[n] - 0.08192f * powf(src[n], 5.f);
140  }
141  break;
142  case ASC_SIN:
143  for (int n = 0; n < nb_samples; n++) {
144  if (FFABS(src[n]) >= M_PI_2)
145  dst[n] = FFSIGN(src[n]);
146  else
147  dst[n] = sinf(src[n]);
148  }
149  break;
150  }
151  }
152 }
153 
155  void **dptr, const void **sptr,
156  int nb_samples, int channels)
157 {
158  double param = s->param;
159 
160  for (int c = 0; c < channels; c++) {
161  const double *src = sptr[c];
162  double *dst = dptr[c];
163 
164  switch (s->type) {
165  case ASC_TANH:
166  for (int n = 0; n < nb_samples; n++) {
167  dst[n] = tanh(src[n] * param);
168  }
169  break;
170  case ASC_ATAN:
171  for (int n = 0; n < nb_samples; n++)
172  dst[n] = 2. / M_PI * atan(src[n] * param);
173  break;
174  case ASC_CUBIC:
175  for (int n = 0; n < nb_samples; n++) {
176  if (FFABS(src[n]) >= 1.5)
177  dst[n] = FFSIGN(src[n]);
178  else
179  dst[n] = src[n] - 0.1481 * pow(src[n], 3.);
180  }
181  break;
182  case ASC_EXP:
183  for (int n = 0; n < nb_samples; n++)
184  dst[n] = 2. / (1. + exp(-2. * src[n])) - 1.;
185  break;
186  case ASC_ALG:
187  for (int n = 0; n < nb_samples; n++)
188  dst[n] = src[n] / (sqrt(param + src[n] * src[n]));
189  break;
190  case ASC_QUINTIC:
191  for (int n = 0; n < nb_samples; n++) {
192  if (FFABS(src[n]) >= 1.25)
193  dst[n] = FFSIGN(src[n]);
194  else
195  dst[n] = src[n] - 0.08192 * pow(src[n], 5.);
196  }
197  break;
198  case ASC_SIN:
199  for (int n = 0; n < nb_samples; n++) {
200  if (FFABS(src[n]) >= M_PI_2)
201  dst[n] = FFSIGN(src[n]);
202  else
203  dst[n] = sin(src[n]);
204  }
205  break;
206  }
207  }
208 }
209 
210 static int config_input(AVFilterLink *inlink)
211 {
212  AVFilterContext *ctx = inlink->dst;
213  ASoftClipContext *s = ctx->priv;
214 
215  switch (inlink->format) {
216  case AV_SAMPLE_FMT_FLT:
217  case AV_SAMPLE_FMT_FLTP: s->filter = filter_flt; break;
218  case AV_SAMPLE_FMT_DBL:
219  case AV_SAMPLE_FMT_DBLP: s->filter = filter_dbl; break;
220  }
221 
222  return 0;
223 }
224 
225 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
226 {
227  AVFilterContext *ctx = inlink->dst;
228  AVFilterLink *outlink = ctx->outputs[0];
229  ASoftClipContext *s = ctx->priv;
230  int nb_samples, channels;
231  AVFrame *out;
232 
233  if (av_frame_is_writable(in)) {
234  out = in;
235  } else {
236  out = ff_get_audio_buffer(outlink, in->nb_samples);
237  if (!out) {
238  av_frame_free(&in);
239  return AVERROR(ENOMEM);
240  }
241  av_frame_copy_props(out, in);
242  }
243 
244  if (av_sample_fmt_is_planar(in->format)) {
245  nb_samples = in->nb_samples;
246  channels = in->channels;
247  } else {
248  nb_samples = in->channels * in->nb_samples;
249  channels = 1;
250  }
251 
252  s->filter(s, (void **)out->extended_data, (const void **)in->extended_data,
253  nb_samples, channels);
254 
255  if (out != in)
256  av_frame_free(&in);
257 
258  return ff_filter_frame(outlink, out);
259 }
260 
261 static const AVFilterPad inputs[] = {
262  {
263  .name = "default",
264  .type = AVMEDIA_TYPE_AUDIO,
265  .filter_frame = filter_frame,
266  .config_props = config_input,
267  },
268  { NULL }
269 };
270 
271 static const AVFilterPad outputs[] = {
272  {
273  .name = "default",
274  .type = AVMEDIA_TYPE_AUDIO,
275  },
276  { NULL }
277 };
278 
280  .name = "asoftclip",
281  .description = NULL_IF_CONFIG_SMALL("Audio Soft Clipper."),
282  .query_formats = query_formats,
283  .priv_size = sizeof(ASoftClipContext),
284  .priv_class = &asoftclip_class,
285  .inputs = inputs,
286  .outputs = outputs,
288 };
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
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
AVOption.
Definition: opt.h:246
static int config_input(AVFilterLink *inlink)
Definition: af_asoftclip.c:210
Main libavfilter public API header.
channels
Definition: aptx.c:30
static const AVOption asoftclip_options[]
Definition: af_asoftclip.c:51
double, planar
Definition: samplefmt.h:70
static const AVFilterPad outputs[]
Definition: af_asoftclip.c:271
static int query_formats(AVFilterContext *ctx)
Definition: af_asoftclip.c:66
#define src
Definition: vp8dsp.c:254
void(* filter)(struct ASoftClipContext *s, void **dst, const void **src, int nb_samples, int channels)
Definition: af_asoftclip.c:44
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
#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
const char * name
Pad name.
Definition: internal.h:60
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1080
#define OFFSET(x)
Definition: af_asoftclip.c:48
AVOptions.
#define f(width, name)
Definition: cbs_vp9.c:255
#define atanf(x)
Definition: libm.h:40
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt)
Check if the sample format is planar.
Definition: samplefmt.c:112
A filter pad used for either input or output.
Definition: internal.h:54
#define expf(x)
Definition: libm.h:283
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
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 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: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
static const AVFilterPad inputs[]
Definition: af_asoftclip.c:261
int8_t exp
Definition: eval.c:72
#define A
Definition: af_asoftclip.c:49
#define powf(x, y)
Definition: libm.h:50
int channels
number of audio channels, only used for audio.
Definition: frame.h:601
audio channel layout utility functions
#define FFSIGN(a)
Definition: common.h:73
#define M_PI_2
Definition: mathematics.h:55
AVFormatContext * ctx
Definition: movenc.c:48
AVFILTER_DEFINE_CLASS(asoftclip)
#define FFABS(a)
Absolute value, Note, INT_MIN / INT64_MIN result in undefined behavior as they are not representable ...
Definition: common.h:72
#define s(width, name)
Definition: cbs_vp9.c:257
int n
Definition: avisynth_c.h:760
A list of supported channel layouts.
Definition: formats.h:85
int format
format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames...
Definition: frame.h:368
#define sinf(x)
Definition: libm.h:419
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
typedef void(RENAME(mix_any_func_type))
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:594
AVFilter ff_af_asoftclip
Definition: af_asoftclip.c:279
uint8_t pi<< 24) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0f/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8, uint8_t,(*(const uint8_t *) pi - 0x80) *(1.0/(1<< 7))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16, int16_t,(*(const int16_t *) pi >> 8)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16, int16_t, *(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32, int32_t,(*(const int32_t *) pi >> 24)+0x80) CONV_FUNC_GROUP(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32, int32_t, *(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, float, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, float, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, float, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, double, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, double, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC_GROUP(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, double, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) #define SET_CONV_FUNC_GROUP(ofmt, ifmt) static void set_generic_function(AudioConvert *ac) { } void ff_audio_convert_free(AudioConvert **ac) { if(! *ac) return;ff_dither_free(&(*ac) ->dc);av_freep(ac);} AudioConvert *ff_audio_convert_alloc(AVAudioResampleContext *avr, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, int sample_rate, int apply_map) { AudioConvert *ac;int in_planar, out_planar;ac=av_mallocz(sizeof(*ac));if(!ac) return NULL;ac->avr=avr;ac->out_fmt=out_fmt;ac->in_fmt=in_fmt;ac->channels=channels;ac->apply_map=apply_map;if(avr->dither_method !=AV_RESAMPLE_DITHER_NONE &&av_get_packed_sample_fmt(out_fmt)==AV_SAMPLE_FMT_S16 &&av_get_bytes_per_sample(in_fmt) > 2) { ac->dc=ff_dither_alloc(avr, out_fmt, in_fmt, channels, sample_rate, apply_map);if(!ac->dc) { av_free(ac);return NULL;} return ac;} in_planar=ff_sample_fmt_is_planar(in_fmt, channels);out_planar=ff_sample_fmt_is_planar(out_fmt, channels);if(in_planar==out_planar) { ac->func_type=CONV_FUNC_TYPE_FLAT;ac->planes=in_planar ? ac->channels :1;} else if(in_planar) ac->func_type=CONV_FUNC_TYPE_INTERLEAVE;else ac->func_type=CONV_FUNC_TYPE_DEINTERLEAVE;set_generic_function(ac);if(ARCH_AARCH64) ff_audio_convert_init_aarch64(ac);if(ARCH_ARM) ff_audio_convert_init_arm(ac);if(ARCH_X86) ff_audio_convert_init_x86(ac);return ac;} int ff_audio_convert(AudioConvert *ac, AudioData *out, AudioData *in) { int use_generic=1;int len=in->nb_samples;int p;if(ac->dc) { av_log(ac->avr, AV_LOG_TRACE, "%d samples - audio_convert: %s to %s (dithered)\", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
const char * name
Filter name.
Definition: avfilter.h:148
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
#define flags(name, subs,...)
Definition: cbs_av1.c:564
ASoftClipTypes
Definition: af_asoftclip.c:27
static void filter_dbl(ASoftClipContext *s, void **dptr, const void **sptr, int nb_samples, int channels)
Definition: af_asoftclip.c:154
static double c[64]
A list of supported formats for one end of a filter link.
Definition: formats.h:64
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_asoftclip.c:225
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:701
FILE * out
Definition: movenc.c:54
#define M_PI
Definition: mathematics.h:52
formats
Definition: signature.h:48
AVFilterChannelLayouts * ff_all_channel_counts(void)
Construct an AVFilterChannelLayouts coding for any channel layout, with known or unknown disposition...
Definition: formats.c:410
static void filter_flt(ASoftClipContext *s, void **dptr, const void **sptr, int nb_samples, int channels)
Definition: af_asoftclip.c:98
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
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