FFmpeg
af_aemphasis.c
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1 /*
2  * Copyright (c) 2001-2010 Krzysztof Foltman, Markus Schmidt, Thor Harald Johansen, Damien Zammit and others
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 
21 #include "libavutil/opt.h"
22 #include "avfilter.h"
23 #include "internal.h"
24 #include "audio.h"
25 
26 typedef struct BiquadCoeffs {
27  double a0, a1, a2, b1, b2;
28 } BiquadCoeffs;
29 
30 typedef struct RIAACurve {
34 } RIAACurve;
35 
36 typedef struct AudioEmphasisContext {
37  const AVClass *class;
38  int mode, type;
39  double level_in, level_out;
40 
42 
45 
46 #define OFFSET(x) offsetof(AudioEmphasisContext, x)
47 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
48 
49 static const AVOption aemphasis_options[] = {
50  { "level_in", "set input gain", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
51  { "level_out", "set output gain", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0, 64, FLAGS },
52  { "mode", "set filter mode", OFFSET(mode), AV_OPT_TYPE_INT, {.i64=0}, 0, 1, FLAGS, "mode" },
53  { "reproduction", NULL, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "mode" },
54  { "production", NULL, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "mode" },
55  { "type", "set filter type", OFFSET(type), AV_OPT_TYPE_INT, {.i64=4}, 0, 8, FLAGS, "type" },
56  { "col", "Columbia", 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, FLAGS, "type" },
57  { "emi", "EMI", 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, FLAGS, "type" },
58  { "bsi", "BSI (78RPM)", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, FLAGS, "type" },
59  { "riaa", "RIAA", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, FLAGS, "type" },
60  { "cd", "Compact Disc (CD)", 0, AV_OPT_TYPE_CONST, {.i64=4}, 0, 0, FLAGS, "type" },
61  { "50fm", "50µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=5}, 0, 0, FLAGS, "type" },
62  { "75fm", "75µs (FM)", 0, AV_OPT_TYPE_CONST, {.i64=6}, 0, 0, FLAGS, "type" },
63  { "50kf", "50µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=7}, 0, 0, FLAGS, "type" },
64  { "75kf", "75µs (FM-KF)", 0, AV_OPT_TYPE_CONST, {.i64=8}, 0, 0, FLAGS, "type" },
65  { NULL }
66 };
67 
68 AVFILTER_DEFINE_CLASS(aemphasis);
69 
70 static inline void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples,
71  double *w, double level_in, double level_out)
72 {
73  const double a0 = bq->a0;
74  const double a1 = bq->a1;
75  const double a2 = bq->a2;
76  const double b1 = bq->b1;
77  const double b2 = bq->b2;
78  double w1 = w[0];
79  double w2 = w[1];
80 
81  for (int i = 0; i < nb_samples; i++) {
82  double n = src[i] * level_in;
83  double tmp = n - w1 * b1 - w2 * b2;
84  double out = tmp * a0 + w1 * a1 + w2 * a2;
85 
86  w2 = w1;
87  w1 = tmp;
88 
89  dst[i] = out * level_out;
90  }
91 
92  w[0] = w1;
93  w[1] = w2;
94 }
95 
96 typedef struct ThreadData {
97  AVFrame *in, *out;
98 } ThreadData;
99 
100 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
101 {
102  AudioEmphasisContext *s = ctx->priv;
103  const double level_out = s->level_out;
104  const double level_in = s->level_in;
105  ThreadData *td = arg;
106  AVFrame *out = td->out;
107  AVFrame *in = td->in;
108  const int start = (in->channels * jobnr) / nb_jobs;
109  const int end = (in->channels * (jobnr+1)) / nb_jobs;
110 
111  for (int ch = start; ch < end; ch++) {
112  const double *src = (const double *)in->extended_data[ch];
113  double *w = (double *)s->w->extended_data[ch];
114  double *dst = (double *)out->extended_data[ch];
115 
116  if (s->rc.use_brickw) {
117  biquad_process(&s->rc.brickw, dst, src, in->nb_samples, w + 2, level_in, 1.);
118  biquad_process(&s->rc.r1, dst, dst, in->nb_samples, w, 1., level_out);
119  } else {
120  biquad_process(&s->rc.r1, dst, src, in->nb_samples, w, level_in, level_out);
121  }
122  }
123 
124  return 0;
125 }
126 
128 {
129  AVFilterContext *ctx = inlink->dst;
130  AVFilterLink *outlink = ctx->outputs[0];
131  ThreadData td;
132  AVFrame *out;
133 
134  if (av_frame_is_writable(in)) {
135  out = in;
136  } else {
137  out = ff_get_audio_buffer(outlink, in->nb_samples);
138  if (!out) {
139  av_frame_free(&in);
140  return AVERROR(ENOMEM);
141  }
142  av_frame_copy_props(out, in);
143  }
144 
145  td.in = in; td.out = out;
146  ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
148 
149  if (in != out)
150  av_frame_free(&in);
151  return ff_filter_frame(outlink, out);
152 }
153 
155 {
158  static const enum AVSampleFormat sample_fmts[] = {
161  };
162  int ret;
163 
164  layouts = ff_all_channel_counts();
165  if (!layouts)
166  return AVERROR(ENOMEM);
167  ret = ff_set_common_channel_layouts(ctx, layouts);
168  if (ret < 0)
169  return ret;
170 
171  formats = ff_make_format_list(sample_fmts);
172  if (!formats)
173  return AVERROR(ENOMEM);
174  ret = ff_set_common_formats(ctx, formats);
175  if (ret < 0)
176  return ret;
177 
178  formats = ff_all_samplerates();
179  if (!formats)
180  return AVERROR(ENOMEM);
181  return ff_set_common_samplerates(ctx, formats);
182 }
183 
184 static inline void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
185 {
186  double A = sqrt(peak);
187  double w0 = freq * 2 * M_PI / sr;
188  double alpha = sin(w0) / (2 * q);
189  double cw0 = cos(w0);
190  double tmp = 2 * sqrt(A) * alpha;
191  double b0 = 0, ib0 = 0;
192 
193  bq->a0 = A*( (A+1) + (A-1)*cw0 + tmp);
194  bq->a1 = -2*A*( (A-1) + (A+1)*cw0);
195  bq->a2 = A*( (A+1) + (A-1)*cw0 - tmp);
196  b0 = (A+1) - (A-1)*cw0 + tmp;
197  bq->b1 = 2*( (A-1) - (A+1)*cw0);
198  bq->b2 = (A+1) - (A-1)*cw0 - tmp;
199 
200  ib0 = 1 / b0;
201  bq->b1 *= ib0;
202  bq->b2 *= ib0;
203  bq->a0 *= ib0;
204  bq->a1 *= ib0;
205  bq->a2 *= ib0;
206 }
207 
208 static inline void set_lp_rbj(BiquadCoeffs *bq, double fc, double q, double sr, double gain)
209 {
210  double omega = 2.0 * M_PI * fc / sr;
211  double sn = sin(omega);
212  double cs = cos(omega);
213  double alpha = sn/(2 * q);
214  double inv = 1.0/(1.0 + alpha);
215 
216  bq->a2 = bq->a0 = gain * inv * (1.0 - cs) * 0.5;
217  bq->a1 = bq->a0 + bq->a0;
218  bq->b1 = (-2.0 * cs * inv);
219  bq->b2 = ((1.0 - alpha) * inv);
220 }
221 
222 static double freq_gain(BiquadCoeffs *c, double freq, double sr)
223 {
224  double zr, zi;
225 
226  freq *= 2.0 * M_PI / sr;
227  zr = cos(freq);
228  zi = -sin(freq);
229 
230  /* |(a0 + a1*z + a2*z^2)/(1 + b1*z + b2*z^2)| */
231  return hypot(c->a0 + c->a1*zr + c->a2*(zr*zr-zi*zi), c->a1*zi + 2*c->a2*zr*zi) /
232  hypot(1 + c->b1*zr + c->b2*(zr*zr-zi*zi), c->b1*zi + 2*c->b2*zr*zi);
233 }
234 
236 {
237  double i, j, k, g, t, a0, a1, a2, b1, b2, tau1, tau2, tau3;
238  double cutfreq, gain1kHz, gc, sr = inlink->sample_rate;
239  AVFilterContext *ctx = inlink->dst;
240  AudioEmphasisContext *s = ctx->priv;
241  BiquadCoeffs coeffs;
242 
243  if (!s->w)
244  s->w = ff_get_audio_buffer(inlink, 4);
245  if (!s->w)
246  return AVERROR(ENOMEM);
247 
248  switch (s->type) {
249  case 0: //"Columbia"
250  i = 100.;
251  j = 500.;
252  k = 1590.;
253  break;
254  case 1: //"EMI"
255  i = 70.;
256  j = 500.;
257  k = 2500.;
258  break;
259  case 2: //"BSI(78rpm)"
260  i = 50.;
261  j = 353.;
262  k = 3180.;
263  break;
264  case 3: //"RIAA"
265  default:
266  tau1 = 0.003180;
267  tau2 = 0.000318;
268  tau3 = 0.000075;
269  i = 1. / (2. * M_PI * tau1);
270  j = 1. / (2. * M_PI * tau2);
271  k = 1. / (2. * M_PI * tau3);
272  break;
273  case 4: //"CD Mastering"
274  tau1 = 0.000050;
275  tau2 = 0.000015;
276  tau3 = 0.0000001;// 1.6MHz out of audible range for null impact
277  i = 1. / (2. * M_PI * tau1);
278  j = 1. / (2. * M_PI * tau2);
279  k = 1. / (2. * M_PI * tau3);
280  break;
281  case 5: //"50µs FM (Europe)"
282  tau1 = 0.000050;
283  tau2 = tau1 / 20;// not used
284  tau3 = tau1 / 50;//
285  i = 1. / (2. * M_PI * tau1);
286  j = 1. / (2. * M_PI * tau2);
287  k = 1. / (2. * M_PI * tau3);
288  break;
289  case 6: //"75µs FM (US)"
290  tau1 = 0.000075;
291  tau2 = tau1 / 20;// not used
292  tau3 = tau1 / 50;//
293  i = 1. / (2. * M_PI * tau1);
294  j = 1. / (2. * M_PI * tau2);
295  k = 1. / (2. * M_PI * tau3);
296  break;
297  }
298 
299  i *= 2 * M_PI;
300  j *= 2 * M_PI;
301  k *= 2 * M_PI;
302 
303  t = 1. / sr;
304 
305  //swap a1 b1, a2 b2
306  if (s->type == 7 || s->type == 8) {
307  double tau = (s->type == 7 ? 0.000050 : 0.000075);
308  double f = 1.0 / (2 * M_PI * tau);
309  double nyq = sr * 0.5;
310  double gain = sqrt(1.0 + nyq * nyq / (f * f)); // gain at Nyquist
311  double cfreq = sqrt((gain - 1.0) * f * f); // frequency
312  double q = 1.0;
313 
314  if (s->type == 8)
315  q = pow((sr / 3269.0) + 19.5, -0.25); // somewhat poor curve-fit
316  if (s->type == 7)
317  q = pow((sr / 4750.0) + 19.5, -0.25);
318  if (s->mode == 0)
319  set_highshelf_rbj(&s->rc.r1, cfreq, q, 1. / gain, sr);
320  else
321  set_highshelf_rbj(&s->rc.r1, cfreq, q, gain, sr);
322  s->rc.use_brickw = 0;
323  } else {
324  s->rc.use_brickw = 1;
325  if (s->mode == 0) { // Reproduction
326  g = 1. / (4.+2.*i*t+2.*k*t+i*k*t*t);
327  a0 = (2.*t+j*t*t)*g;
328  a1 = (2.*j*t*t)*g;
329  a2 = (-2.*t+j*t*t)*g;
330  b1 = (-8.+2.*i*k*t*t)*g;
331  b2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
332  } else { // Production
333  g = 1. / (2.*t+j*t*t);
334  a0 = (4.+2.*i*t+2.*k*t+i*k*t*t)*g;
335  a1 = (-8.+2.*i*k*t*t)*g;
336  a2 = (4.-2.*i*t-2.*k*t+i*k*t*t)*g;
337  b1 = (2.*j*t*t)*g;
338  b2 = (-2.*t+j*t*t)*g;
339  }
340 
341  coeffs.a0 = a0;
342  coeffs.a1 = a1;
343  coeffs.a2 = a2;
344  coeffs.b1 = b1;
345  coeffs.b2 = b2;
346 
347  // the coeffs above give non-normalized value, so it should be normalized to produce 0dB at 1 kHz
348  // find actual gain
349  // Note: for FM emphasis, use 100 Hz for normalization instead
350  gain1kHz = freq_gain(&coeffs, 1000.0, sr);
351  // divide one filter's x[n-m] coefficients by that value
352  gc = 1.0 / gain1kHz;
353  s->rc.r1.a0 = coeffs.a0 * gc;
354  s->rc.r1.a1 = coeffs.a1 * gc;
355  s->rc.r1.a2 = coeffs.a2 * gc;
356  s->rc.r1.b1 = coeffs.b1;
357  s->rc.r1.b2 = coeffs.b2;
358  }
359 
360  cutfreq = FFMIN(0.45 * sr, 21000.);
361  set_lp_rbj(&s->rc.brickw, cutfreq, 0.707, sr, 1.);
362 
363  return 0;
364 }
365 
366 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
367  char *res, int res_len, int flags)
368 {
369  int ret;
370 
371  ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
372  if (ret < 0)
373  return ret;
374 
375  return config_input(ctx->inputs[0]);
376 }
377 
379 {
380  AudioEmphasisContext *s = ctx->priv;
381 
382  av_frame_free(&s->w);
383 }
384 
386  {
387  .name = "default",
388  .type = AVMEDIA_TYPE_AUDIO,
389  .config_props = config_input,
390  .filter_frame = filter_frame,
391  },
392  { NULL }
393 };
394 
396  {
397  .name = "default",
398  .type = AVMEDIA_TYPE_AUDIO,
399  },
400  { NULL }
401 };
402 
404  .name = "aemphasis",
405  .description = NULL_IF_CONFIG_SMALL("Audio emphasis."),
406  .priv_size = sizeof(AudioEmphasisContext),
407  .priv_class = &aemphasis_class,
408  .uninit = uninit,
410  .inputs = avfilter_af_aemphasis_inputs,
411  .outputs = avfilter_af_aemphasis_outputs,
415 };
#define NULL
Definition: coverity.c:32
AVFrame * out
Definition: af_adeclick.c:502
This structure describes decoded (raw) audio or video data.
Definition: frame.h:318
AVOption.
Definition: opt.h:248
static void biquad_process(BiquadCoeffs *bq, double *dst, const double *src, int nb_samples, double *w, double level_in, double level_out)
Definition: af_aemphasis.c:70
Main libavfilter public API header.
const char * g
Definition: vf_curves.c:117
int use_brickw
Definition: af_aemphasis.c:33
double, planar
Definition: samplefmt.h:70
GLint GLenum type
Definition: opengl_enc.c:104
int ff_set_common_channel_layouts(AVFilterContext *ctx, AVFilterChannelLayouts *channel_layouts)
A helper for query_formats() which sets all links to the same list of channel layouts/sample rates...
Definition: formats.c:569
static const AVOption aemphasis_options[]
Definition: af_aemphasis.c:49
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:287
#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:126
static double freq_gain(BiquadCoeffs *c, double freq, double sr)
Definition: af_aemphasis.c:222
const char * name
Pad name.
Definition: internal.h:60
AVFilterLink ** inputs
array of pointers to input links
Definition: avfilter.h:349
int ff_filter_frame(AVFilterLink *link, AVFrame *frame)
Send a frame of data to the next filter.
Definition: avfilter.c:1094
#define av_cold
Definition: attributes.h:88
AVOptions.
#define OFFSET(x)
Definition: af_aemphasis.c:46
#define f(width, name)
Definition: cbs_vp9.c:255
Undefined Behavior In the C some operations are like signed integer dereferencing freed accessing outside allocated Undefined Behavior must not occur in a C it is not safe even if the output of undefined operations is unused The unsafety may seem nit picking but Optimizing compilers have in fact optimized code on the assumption that no undefined Behavior occurs Optimizing code based on wrong assumptions can and has in some cases lead to effects beyond the output of computations The signed integer overflow problem in speed critical code Code which is highly optimized and works with signed integers sometimes has the problem that often the output of the computation does not c
Definition: undefined.txt:32
AVFILTER_DEFINE_CLASS(aemphasis)
BiquadCoeffs brickw
Definition: af_aemphasis.c:32
BiquadCoeffs r1
Definition: af_aemphasis.c:31
AVFilter ff_af_aemphasis
Definition: af_aemphasis.c:403
#define A(x)
Definition: vp56_arith.h:28
A filter pad used for either input or output.
Definition: internal.h:54
#define src
Definition: vp8dsp.c:255
#define fc(width, name, range_min, range_max)
Definition: cbs_av1.c:551
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:588
#define td
Definition: regdef.h:70
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:204
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:117
int ff_filter_process_command(AVFilterContext *ctx, const char *cmd, const char *arg, char *res, int res_len, int flags)
Generic processing of user supplied commands that are set in the same way as the filter options...
Definition: avfilter.c:882
void * priv
private data for use by the filter
Definition: avfilter.h:356
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:117
const char * arg
Definition: jacosubdec.c:66
static av_const double hypot(double x, double y)
Definition: libm.h:366
int channels
number of audio channels, only used for audio.
Definition: frame.h:624
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
int ff_filter_get_nb_threads(AVFilterContext *ctx)
Get number of threads for current filter instance.
Definition: avfilter.c:802
static void set_lp_rbj(BiquadCoeffs *bq, double fc, double q, double sr, double gain)
Definition: af_aemphasis.c:208
#define FFMIN(a, b)
Definition: common.h:105
uint8_t w
Definition: llviddspenc.c:39
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_aemphasis.c:366
AVFormatContext * ctx
Definition: movenc.c:48
static const AVFilterPad avfilter_af_aemphasis_inputs[]
Definition: af_aemphasis.c:385
#define s(width, name)
Definition: cbs_vp9.c:257
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
static int query_formats(AVFilterContext *ctx)
Definition: af_aemphasis.c:154
A list of supported channel layouts.
Definition: formats.h:86
if(ret)
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int av_frame_is_writable(AVFrame *frame)
Check if the frame data is writable.
Definition: frame.c:595
Used for passing data between threads.
Definition: dsddec.c:67
static const int16_t alpha[]
Definition: ilbcdata.h:55
static const AVFilterPad avfilter_af_aemphasis_outputs[]
Definition: af_aemphasis.c:395
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)\n", len, av_get_sample_fmt_name(ac->in_fmt), av_get_sample_fmt_name(ac->out_fmt));return ff_convert_dither(ac-> in
static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
Definition: af_aemphasis.c:100
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:145
const char * name
Filter name.
Definition: avfilter.h:149
AVFilterLink ** outputs
array of pointers to output links
Definition: avfilter.h:353
enum MovChannelLayoutTag * layouts
Definition: mov_chan.c:434
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:422
static void set_highshelf_rbj(BiquadCoeffs *bq, double freq, double q, double peak, double sr)
Definition: af_aemphasis.c:184
#define flags(name, subs,...)
Definition: cbs_av1.c:561
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
Definition: avfilter.h:381
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
avfilter_execute_func * execute
Definition: internal.h:136
#define FLAGS
Definition: af_aemphasis.c:47
A list of supported formats for one end of a filter link.
Definition: formats.h:65
An instance of a filter.
Definition: avfilter.h:341
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:941
FILE * out
Definition: movenc.c:54
#define M_PI
Definition: mathematics.h:52
AVFrame * in
Definition: af_adenorm.c:223
formats
Definition: signature.h:48
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:437
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:365
static int config_input(AVFilterLink *inlink)
Definition: af_aemphasis.c:235
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_aemphasis.c:378
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:384
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:576
int av_frame_copy_props(AVFrame *dst, const AVFrame *src)
Copy only "metadata" fields from src to dst.
Definition: frame.c:659
int i
Definition: input.c:407
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_aemphasis.c:127
static uint8_t tmp[11]
Definition: aes_ctr.c:27