FFmpeg  4.0
af_stereotools.c
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1 /*
2  * Copyright (C) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen
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 
27 typedef struct StereoToolsContext {
28  const AVClass *class;
29 
30  int softclip;
31  int mute_l;
32  int mute_r;
33  int phase_l;
34  int phase_r;
35  int mode;
36  int bmode_in;
37  int bmode_out;
38  double slev;
39  double sbal;
40  double mlev;
41  double mpan;
42  double phase;
43  double base;
44  double delay;
45  double balance_in;
46  double balance_out;
49  double sc_level;
51  double level_in;
52  double level_out;
53 
54  double *buffer;
55  int length;
56  int pos;
58 
59 #define OFFSET(x) offsetof(StereoToolsContext, x)
60 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
61 
62 static const AVOption stereotools_options[] = {
63  { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
64  { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
65  { "balance_in", "set balance in", OFFSET(balance_in), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
66  { "balance_out", "set balance out", OFFSET(balance_out), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
67  { "softclip", "enable softclip", OFFSET(softclip), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
68  { "mutel", "mute L", OFFSET(mute_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
69  { "muter", "mute R", OFFSET(mute_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
70  { "phasel", "phase L", OFFSET(phase_l), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
71  { "phaser", "phase R", OFFSET(phase_r), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
72  { "mode", "set stereo mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 8, A, "mode" },
73  { "lr>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "mode" },
74  { "lr>ms", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "mode" },
75  { "ms>lr", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "mode" },
76  { "lr>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "mode" },
77  { "lr>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, A, "mode" },
78  { "lr>l+r", 0, 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, A, "mode" },
79  { "lr>rl", 0, 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, A, "mode" },
80  { "ms>ll", 0, 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, A, "mode" },
81  { "ms>rr", 0, 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, A, "mode" },
82  { "slev", "set side level", OFFSET(slev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
83  { "sbal", "set side balance", OFFSET(sbal), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
84  { "mlev", "set middle level", OFFSET(mlev), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
85  { "mpan", "set middle pan", OFFSET(mpan), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
86  { "base", "set stereo base", OFFSET(base), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
87  { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -20, 20, A },
88  { "sclevel", "set S/C level", OFFSET(sc_level), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 1, 100, A },
89  { "phase", "set stereo phase", OFFSET(phase), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 360, A },
90  { "bmode_in", "set balance in mode", OFFSET(bmode_in), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
91  { "balance", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "bmode" },
92  { "amplitude", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "bmode" },
93  { "power", 0, 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "bmode" },
94  { "bmode_out", "set balance out mode", OFFSET(bmode_out), AV_OPT_TYPE_INT, {.i64=0}, 0, 2, A, "bmode" },
95  { NULL }
96 };
97 
98 AVFILTER_DEFINE_CLASS(stereotools);
99 
101 {
104  int ret;
105 
106  if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
107  (ret = ff_set_common_formats (ctx , formats )) < 0 ||
108  (ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO)) < 0 ||
109  (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
110  return ret;
111 
112  formats = ff_all_samplerates();
113  return ff_set_common_samplerates(ctx, formats);
114 }
115 
116 static int config_input(AVFilterLink *inlink)
117 {
118  AVFilterContext *ctx = inlink->dst;
119  StereoToolsContext *s = ctx->priv;
120 
121  s->length = 2 * inlink->sample_rate * 0.05;
122  if (s->length <= 1 || s->length & 1) {
123  av_log(ctx, AV_LOG_ERROR, "sample rate is too small\n");
124  return AVERROR(EINVAL);
125  }
126  s->buffer = av_calloc(s->length, sizeof(*s->buffer));
127  if (!s->buffer)
128  return AVERROR(ENOMEM);
129 
130  s->inv_atan_shape = 1.0 / atan(s->sc_level);
131  s->phase_cos_coef = cos(s->phase / 180 * M_PI);
132  s->phase_sin_coef = sin(s->phase / 180 * M_PI);
133 
134  return 0;
135 }
136 
137 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
138 {
139  AVFilterContext *ctx = inlink->dst;
140  AVFilterLink *outlink = ctx->outputs[0];
141  StereoToolsContext *s = ctx->priv;
142  const double *src = (const double *)in->data[0];
143  const double sb = s->base < 0 ? s->base * 0.5 : s->base;
144  const double sbal = 1 + s->sbal;
145  const double mpan = 1 + s->mpan;
146  const double slev = s->slev;
147  const double mlev = s->mlev;
148  const double balance_in = s->balance_in;
149  const double balance_out = s->balance_out;
150  const double level_in = s->level_in;
151  const double level_out = s->level_out;
152  const double sc_level = s->sc_level;
153  const double delay = s->delay;
154  const int length = s->length;
155  const int mute_l = s->mute_l;
156  const int mute_r = s->mute_r;
157  const int phase_l = s->phase_l;
158  const int phase_r = s->phase_r;
159  double *buffer = s->buffer;
160  AVFrame *out;
161  double *dst;
162  int nbuf = inlink->sample_rate * (fabs(delay) / 1000.);
163  int n;
164 
165  nbuf -= nbuf % 2;
166  if (av_frame_is_writable(in)) {
167  out = in;
168  } else {
169  out = ff_get_audio_buffer(outlink, in->nb_samples);
170  if (!out) {
171  av_frame_free(&in);
172  return AVERROR(ENOMEM);
173  }
175  }
176  dst = (double *)out->data[0];
177 
178  for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
179  double L = src[0], R = src[1], l, r, m, S, gl, gr, gd;
180 
181  L *= level_in;
182  R *= level_in;
183 
184  gl = 1. - FFMAX(0., balance_in);
185  gr = 1. + FFMIN(0., balance_in);
186  switch (s->bmode_in) {
187  case 1:
188  gd = gl - gr;
189  gl = 1. + gd;
190  gr = 1. - gd;
191  break;
192  case 2:
193  if (balance_in < 0.) {
194  gr = FFMAX(0.5, gr);
195  gl = 1. / gr;
196  } else if (balance_in > 0.) {
197  gl = FFMAX(0.5, gl);
198  gr = 1. / gl;
199  }
200  break;
201  }
202  L *= gl;
203  R *= gr;
204 
205  if (s->softclip) {
206  R = s->inv_atan_shape * atan(R * sc_level);
207  L = s->inv_atan_shape * atan(L * sc_level);
208  }
209 
210  switch (s->mode) {
211  case 0:
212  m = (L + R) * 0.5;
213  S = (L - R) * 0.5;
214  l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
215  r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
216  L = l;
217  R = r;
218  break;
219  case 1:
220  l = L * FFMIN(1., 2. - sbal);
221  r = R * FFMIN(1., sbal);
222  L = 0.5 * (l + r) * mlev;
223  R = 0.5 * (l - r) * slev;
224  break;
225  case 2:
226  l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
227  r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
228  L = l;
229  R = r;
230  break;
231  case 3:
232  R = L;
233  break;
234  case 4:
235  L = R;
236  break;
237  case 5:
238  L = (L + R) / 2;
239  R = L;
240  break;
241  case 6:
242  l = L;
243  L = R;
244  R = l;
245  m = (L + R) * 0.5;
246  S = (L - R) * 0.5;
247  l = m * mlev * FFMIN(1., 2. - mpan) + S * slev * FFMIN(1., 2. - sbal);
248  r = m * mlev * FFMIN(1., mpan) - S * slev * FFMIN(1., sbal);
249  L = l;
250  R = r;
251  break;
252  case 7:
253  l = L * mlev * FFMIN(1., 2. - mpan) + R * slev * FFMIN(1., 2. - sbal);
254  L = l;
255  R = l;
256  break;
257  case 8:
258  r = L * mlev * FFMIN(1., mpan) - R * slev * FFMIN(1., sbal);
259  L = r;
260  R = r;
261  break;
262  }
263 
264  L *= 1. - mute_l;
265  R *= 1. - mute_r;
266 
267  L *= (2. * (1. - phase_l)) - 1.;
268  R *= (2. * (1. - phase_r)) - 1.;
269 
270  buffer[s->pos ] = L;
271  buffer[s->pos+1] = R;
272 
273  if (delay > 0.) {
274  R = buffer[(s->pos - (int)nbuf + 1 + length) % length];
275  } else if (delay < 0.) {
276  L = buffer[(s->pos - (int)nbuf + length) % length];
277  }
278 
279  l = L + sb * L - sb * R;
280  r = R + sb * R - sb * L;
281 
282  L = l;
283  R = r;
284 
285  l = L * s->phase_cos_coef - R * s->phase_sin_coef;
286  r = L * s->phase_sin_coef + R * s->phase_cos_coef;
287 
288  L = l;
289  R = r;
290 
291  s->pos = (s->pos + 2) % s->length;
292 
293  gl = 1. - FFMAX(0., balance_out);
294  gr = 1. + FFMIN(0., balance_out);
295  switch (s->bmode_out) {
296  case 1:
297  gd = gl - gr;
298  gl = 1. + gd;
299  gr = 1. - gd;
300  break;
301  case 2:
302  if (balance_out < 0.) {
303  gr = FFMAX(0.5, gr);
304  gl = 1. / gr;
305  } else if (balance_out > 0.) {
306  gl = FFMAX(0.5, gl);
307  gr = 1. / gl;
308  }
309  break;
310  }
311  L *= gl;
312  R *= gr;
313 
314 
315  L *= level_out;
316  R *= level_out;
317 
318  dst[0] = L;
319  dst[1] = R;
320  }
321 
322  if (out != in)
323  av_frame_free(&in);
324  return ff_filter_frame(outlink, out);
325 }
326 
328 {
329  StereoToolsContext *s = ctx->priv;
330 
331  av_freep(&s->buffer);
332 }
333 
334 static const AVFilterPad inputs[] = {
335  {
336  .name = "default",
337  .type = AVMEDIA_TYPE_AUDIO,
338  .filter_frame = filter_frame,
339  .config_props = config_input,
340  },
341  { NULL }
342 };
343 
344 static const AVFilterPad outputs[] = {
345  {
346  .name = "default",
347  .type = AVMEDIA_TYPE_AUDIO,
348  },
349  { NULL }
350 };
351 
353  .name = "stereotools",
354  .description = NULL_IF_CONFIG_SMALL("Apply various stereo tools."),
355  .query_formats = query_formats,
356  .priv_size = sizeof(StereoToolsContext),
357  .priv_class = &stereotools_class,
358  .uninit = uninit,
359  .inputs = inputs,
360  .outputs = outputs,
361 };
static const AVOption stereotools_options[]
#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
const char * s
Definition: avisynth_c.h:768
This structure describes decoded (raw) audio or video data.
Definition: frame.h:218
AVOption.
Definition: opt.h:246
Main libavfilter public API header.
#define AV_CH_LAYOUT_STEREO
#define src
Definition: vp8dsp.c:254
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
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 av_cold
Definition: attributes.h:82
AVOptions.
#define OFFSET(x)
#define av_log(a,...)
#define A
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
A filter pad used for either input or output.
Definition: internal.h:54
#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
#define R
Definition: huffyuvdsp.h:34
int ff_add_channel_layout(AVFilterChannelLayouts **l, uint64_t channel_layout)
Definition: formats.c:343
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 S(s, c, i)
#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
const char * r
Definition: vf_curves.c:111
void * priv
private data for use by the filter
Definition: avfilter.h:353
int ff_add_format(AVFilterFormats **avff, int64_t fmt)
Add fmt to the list of media formats contained in *avff.
Definition: formats.c:337
#define FFMAX(a, b)
Definition: common.h:94
AVFilter ff_af_stereotools
audio channel layout utility functions
static av_cold void uninit(AVFilterContext *ctx)
#define FFMIN(a, b)
Definition: common.h:96
AVFILTER_DEFINE_CLASS(stereotools)
AVFormatContext * ctx
Definition: movenc.c:48
static int query_formats(AVFilterContext *ctx)
int n
Definition: avisynth_c.h:684
#define L(x)
Definition: vp56_arith.h:36
static const AVFilterPad outputs[]
static int config_input(AVFilterLink *inlink)
A list of supported channel layouts.
Definition: formats.h:85
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:592
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
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:395
uint8_t * data[AV_NUM_DATA_POINTERS]
pointer to the picture/channel planes.
Definition: frame.h:232
static const AVFilterPad inputs[]
int
A list of supported formats for one end of a filter link.
Definition: formats.h:64
uint64_t layout
An instance of a filter.
Definition: avfilter.h:338
FILE * out
Definition: movenc.c:54
#define av_freep(p)
#define M_PI
Definition: mathematics.h:52
formats
Definition: signature.h:48
mode
Use these values in ebur128_init (or&#39;ed).
Definition: ebur128.h:83
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:284
for(j=16;j >0;--j)
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:652