FFmpeg
af_superequalizer.c
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1 /*
2  * Copyright (c) 2002 Naoki Shibata
3  * Copyright (c) 2017 Paul B Mahol
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 #include "libavutil/opt.h"
23 
24 #include "libavcodec/avfft.h"
25 
26 #include "audio.h"
27 #include "avfilter.h"
28 #include "filters.h"
29 #include "internal.h"
30 
31 #define NBANDS 17
32 #define M 15
33 
34 typedef struct EqParameter {
35  float lower, upper, gain;
36 } EqParameter;
37 
38 typedef struct SuperEqualizerContext {
39  const AVClass *class;
40 
42 
43  float gains[NBANDS + 1];
44 
45  float fact[M + 1];
46  float aa;
47  float iza;
48  float *ires, *irest;
49  float *fsamples;
50  int winlen, tabsize;
51 
53  RDFTContext *rdft, *irdft;
55 
56 static const float bands[] = {
57  65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
58  1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
59 };
60 
61 static float izero(SuperEqualizerContext *s, float x)
62 {
63  float ret = 1;
64  int m;
65 
66  for (m = 1; m <= M; m++) {
67  float t;
68 
69  t = pow(x / 2, m) / s->fact[m];
70  ret += t*t;
71  }
72 
73  return ret;
74 }
75 
76 static float hn_lpf(int n, float f, float fs)
77 {
78  float t = 1 / fs;
79  float omega = 2 * M_PI * f;
80 
81  if (n * omega * t == 0)
82  return 2 * f * t;
83  return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
84 }
85 
86 static float hn_imp(int n)
87 {
88  return n == 0 ? 1.f : 0.f;
89 }
90 
91 static float hn(int n, EqParameter *param, float fs)
92 {
93  float ret, lhn;
94  int i;
95 
96  lhn = hn_lpf(n, param[0].upper, fs);
97  ret = param[0].gain*lhn;
98 
99  for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
100  float lhn2 = hn_lpf(n, param[i].upper, fs);
101  ret += param[i].gain * (lhn2 - lhn);
102  lhn = lhn2;
103  }
104 
105  ret += param[i].gain * (hn_imp(n) - lhn);
106 
107  return ret;
108 }
109 
110 static float alpha(float a)
111 {
112  if (a <= 21)
113  return 0;
114  if (a <= 50)
115  return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
116  return .1102f * (a - 8.7f);
117 }
118 
119 static float win(SuperEqualizerContext *s, float n, int N)
120 {
121  return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
122 }
123 
124 static void process_param(float *bc, EqParameter *param, float fs)
125 {
126  int i;
127 
128  for (i = 0; i <= NBANDS; i++) {
129  param[i].lower = i == 0 ? 0 : bands[i - 1];
130  param[i].upper = i == NBANDS ? fs : bands[i];
131  param[i].gain = bc[i];
132  }
133 }
134 
135 static int equ_init(SuperEqualizerContext *s, int wb)
136 {
137  int i,j;
138 
139  s->rdft = av_rdft_init(wb, DFT_R2C);
140  s->irdft = av_rdft_init(wb, IDFT_C2R);
141  if (!s->rdft || !s->irdft)
142  return AVERROR(ENOMEM);
143 
144  s->aa = 96;
145  s->winlen = (1 << (wb-1))-1;
146  s->tabsize = 1 << wb;
147 
148  s->ires = av_calloc(s->tabsize, sizeof(float));
149  s->irest = av_calloc(s->tabsize, sizeof(float));
150  s->fsamples = av_calloc(s->tabsize, sizeof(float));
151 
152  for (i = 0; i <= M; i++) {
153  s->fact[i] = 1;
154  for (j = 1; j <= i; j++)
155  s->fact[i] *= j;
156  }
157 
158  s->iza = izero(s, alpha(s->aa));
159 
160  return 0;
161 }
162 
163 static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
164 {
165  const int winlen = s->winlen;
166  const int tabsize = s->tabsize;
167  float *nires;
168  int i;
169 
170  if (fs <= 0)
171  return;
172 
173  process_param(lbc, param, fs);
174  for (i = 0; i < winlen; i++)
175  s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
176  for (; i < tabsize; i++)
177  s->irest[i] = 0;
178 
179  av_rdft_calc(s->rdft, s->irest);
180  nires = s->ires;
181  for (i = 0; i < tabsize; i++)
182  nires[i] = s->irest[i];
183 }
184 
186 {
187  AVFilterContext *ctx = inlink->dst;
188  SuperEqualizerContext *s = ctx->priv;
189  AVFilterLink *outlink = ctx->outputs[0];
190  const float *ires = s->ires;
191  float *fsamples = s->fsamples;
192  int ch, i;
193 
194  AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
195  float *src, *dst, *ptr;
196 
197  if (!out) {
198  av_frame_free(&in);
199  return AVERROR(ENOMEM);
200  }
201 
202  for (ch = 0; ch < in->channels; ch++) {
203  ptr = (float *)out->extended_data[ch];
204  dst = (float *)s->out->extended_data[ch];
205  src = (float *)in->extended_data[ch];
206 
207  for (i = 0; i < in->nb_samples; i++)
208  fsamples[i] = src[i];
209  for (; i < s->tabsize; i++)
210  fsamples[i] = 0;
211 
212  av_rdft_calc(s->rdft, fsamples);
213 
214  fsamples[0] = ires[0] * fsamples[0];
215  fsamples[1] = ires[1] * fsamples[1];
216  for (i = 1; i < s->tabsize / 2; i++) {
217  float re, im;
218 
219  re = ires[i*2 ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
220  im = ires[i*2+1] * fsamples[i*2] + ires[i*2 ] * fsamples[i*2+1];
221 
222  fsamples[i*2 ] = re;
223  fsamples[i*2+1] = im;
224  }
225 
226  av_rdft_calc(s->irdft, fsamples);
227 
228  for (i = 0; i < s->winlen; i++)
229  dst[i] += fsamples[i] / s->tabsize * 2;
230  for (i = s->winlen; i < s->tabsize; i++)
231  dst[i] = fsamples[i] / s->tabsize * 2;
232  for (i = 0; i < s->winlen; i++)
233  ptr[i] = dst[i];
234  for (i = 0; i < s->winlen; i++)
235  dst[i] = dst[i+s->winlen];
236  }
237 
238  out->pts = in->pts;
239  av_frame_free(&in);
240 
241  return ff_filter_frame(outlink, out);
242 }
243 
245 {
246  AVFilterLink *inlink = ctx->inputs[0];
247  AVFilterLink *outlink = ctx->outputs[0];
248  SuperEqualizerContext *s = ctx->priv;
249  AVFrame *in = NULL;
250  int ret;
251 
252  FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
253 
254  ret = ff_inlink_consume_samples(inlink, s->winlen, s->winlen, &in);
255  if (ret < 0)
256  return ret;
257  if (ret > 0)
258  return filter_frame(inlink, in);
259 
260  FF_FILTER_FORWARD_STATUS(inlink, outlink);
261  FF_FILTER_FORWARD_WANTED(outlink, inlink);
262 
263  return FFERROR_NOT_READY;
264 }
265 
267 {
268  SuperEqualizerContext *s = ctx->priv;
269 
270  return equ_init(s, 14);
271 }
272 
274 {
277  static const enum AVSampleFormat sample_fmts[] = {
280  };
281  int ret;
282 
283  layouts = ff_all_channel_counts();
284  if (!layouts)
285  return AVERROR(ENOMEM);
286  ret = ff_set_common_channel_layouts(ctx, layouts);
287  if (ret < 0)
288  return ret;
289 
290  formats = ff_make_format_list(sample_fmts);
291  if ((ret = ff_set_common_formats(ctx, formats)) < 0)
292  return ret;
293 
294  formats = ff_all_samplerates();
295  return ff_set_common_samplerates(ctx, formats);
296 }
297 
299 {
300  AVFilterContext *ctx = inlink->dst;
301  SuperEqualizerContext *s = ctx->priv;
302 
303  s->out = ff_get_audio_buffer(inlink, s->tabsize);
304  if (!s->out)
305  return AVERROR(ENOMEM);
306 
307  return 0;
308 }
309 
310 static int config_output(AVFilterLink *outlink)
311 {
312  AVFilterContext *ctx = outlink->src;
313  SuperEqualizerContext *s = ctx->priv;
314 
315  make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
316 
317  return 0;
318 }
319 
321 {
322  SuperEqualizerContext *s = ctx->priv;
323 
324  av_frame_free(&s->out);
325  av_freep(&s->irest);
326  av_freep(&s->ires);
327  av_freep(&s->fsamples);
328  av_rdft_end(s->rdft);
329  av_rdft_end(s->irdft);
330 }
331 
333  {
334  .name = "default",
335  .type = AVMEDIA_TYPE_AUDIO,
336  .config_props = config_input,
337  },
338  { NULL }
339 };
340 
342  {
343  .name = "default",
344  .type = AVMEDIA_TYPE_AUDIO,
345  .config_props = config_output,
346  },
347  { NULL }
348 };
349 
350 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
351 #define OFFSET(x) offsetof(SuperEqualizerContext, x)
352 
354  { "1b", "set 65Hz band gain", OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
355  { "2b", "set 92Hz band gain", OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
356  { "3b", "set 131Hz band gain", OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
357  { "4b", "set 185Hz band gain", OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
358  { "5b", "set 262Hz band gain", OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
359  { "6b", "set 370Hz band gain", OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
360  { "7b", "set 523Hz band gain", OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
361  { "8b", "set 740Hz band gain", OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
362  { "9b", "set 1047Hz band gain", OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
363  { "10b", "set 1480Hz band gain", OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
364  { "11b", "set 2093Hz band gain", OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
365  { "12b", "set 2960Hz band gain", OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
366  { "13b", "set 4186Hz band gain", OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
367  { "14b", "set 5920Hz band gain", OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
368  { "15b", "set 8372Hz band gain", OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
369  { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
370  { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
371  { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
372  { NULL }
373 };
374 
375 AVFILTER_DEFINE_CLASS(superequalizer);
376 
378  .name = "superequalizer",
379  .description = NULL_IF_CONFIG_SMALL("Apply 18 band equalization filter."),
380  .priv_size = sizeof(SuperEqualizerContext),
381  .priv_class = &superequalizer_class,
383  .init = init,
384  .activate = activate,
385  .uninit = uninit,
386  .inputs = superequalizer_inputs,
387  .outputs = superequalizer_outputs,
388 };
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
static float alpha(float a)
This structure describes decoded (raw) audio or video data.
Definition: frame.h:295
static int query_formats(AVFilterContext *ctx)
static const AVFilterPad superequalizer_outputs[]
AVOption.
Definition: opt.h:246
static const AVOption superequalizer_options[]
float re
Definition: fft.c:82
#define NBANDS
Main libavfilter public API header.
static float win(SuperEqualizerContext *s, float n, int N)
FF_FILTER_FORWARD_STATUS(inlink, outlink)
uint8_t pi<< 24) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_U8,(uint64_t)((*(const uint8_t *) pi-0x80U))<< 56) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0f/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_U8,(*(const uint8_t *) pi-0x80)*(1.0/(1<< 7))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S16,(*(const int16_t *) pi >>8)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S16,(uint64_t)(*(const int16_t *) pi)<< 48) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0f/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S16,*(const int16_t *) pi *(1.0/(1<< 15))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S32,(*(const int32_t *) pi >>24)+0x80) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_S32,(uint64_t)(*(const int32_t *) pi)<< 32) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0f/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S32,*(const int32_t *) pi *(1.0/(1U<< 31))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_S64,(*(const int64_t *) pi >>56)+0x80) CONV_FUNC(AV_SAMPLE_FMT_FLT, float, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0f/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64,*(const int64_t *) pi *(1.0/(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8(lrintf(*(const float *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16(lrintf(*(const float *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_FLT, llrintf(*(const float *) pi *(INT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_U8, uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8(lrint(*(const double *) pi *(1<< 7))+0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16(lrint(*(const double *) pi *(1<< 15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double *) pi *(1U<< 31)))) CONV_FUNC(AV_SAMPLE_FMT_S64, int64_t, AV_SAMPLE_FMT_DBL, llrint(*(const double *) pi *(INT64_C(1)<< 63)))#define FMT_PAIR_FUNC(out, in) static conv_func_type *const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB *AV_SAMPLE_FMT_NB]={FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_U8), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S16), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S32), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_FLT), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_DBL), FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_S64), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S64, AV_SAMPLE_FMT_S64),};static void cpy1(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, len);}static void cpy2(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 2 *len);}static void cpy4(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 4 *len);}static void cpy8(uint8_t **dst, const uint8_t **src, int len){memcpy(*dst,*src, 8 *len);}AudioConvert *swri_audio_convert_alloc(enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt, int channels, const int *ch_map, int flags){AudioConvert *ctx;conv_func_type *f=fmt_pair_to_conv_functions[av_get_packed_sample_fmt(out_fmt)+AV_SAMPLE_FMT_NB *av_get_packed_sample_fmt(in_fmt)];if(!f) return NULL;ctx=av_mallocz(sizeof(*ctx));if(!ctx) return NULL;if(channels==1){in_fmt=av_get_planar_sample_fmt(in_fmt);out_fmt=av_get_planar_sample_fmt(out_fmt);}ctx->channels=channels;ctx->conv_f=f;ctx->ch_map=ch_map;if(in_fmt==AV_SAMPLE_FMT_U8||in_fmt==AV_SAMPLE_FMT_U8P) memset(ctx->silence, 0x80, sizeof(ctx->silence));if(out_fmt==in_fmt &&!ch_map){switch(av_get_bytes_per_sample(in_fmt)){case 1:ctx->simd_f=cpy1;break;case 2:ctx->simd_f=cpy2;break;case 4:ctx->simd_f=cpy4;break;case 8:ctx->simd_f=cpy8;break;}}if(HAVE_X86ASM &&1) swri_audio_convert_init_x86(ctx, out_fmt, in_fmt, channels);if(ARCH_ARM) swri_audio_convert_init_arm(ctx, out_fmt, in_fmt, channels);if(ARCH_AARCH64) swri_audio_convert_init_aarch64(ctx, out_fmt, in_fmt, channels);return ctx;}void swri_audio_convert_free(AudioConvert **ctx){av_freep(ctx);}int swri_audio_convert(AudioConvert *ctx, AudioData *out, AudioData *in, int len){int ch;int off=0;const int os=(out->planar?1:out->ch_count)*out->bps;unsigned misaligned=0;av_assert0(ctx->channels==out->ch_count);if(ctx->in_simd_align_mask){int planes=in->planar?in->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) in->ch[ch];misaligned|=m &ctx->in_simd_align_mask;}if(ctx->out_simd_align_mask){int planes=out->planar?out->ch_count:1;unsigned m=0;for(ch=0;ch< planes;ch++) m|=(intptr_t) out->ch[ch];misaligned|=m &ctx->out_simd_align_mask;}if(ctx->simd_f &&!ctx->ch_map &&!misaligned){off=len &~15;av_assert1(off >=0);av_assert1(off<=len);av_assert2(ctx->channels==SWR_CH_MAX||!in->ch[ctx->channels]);if(off >0){if(out->planar==in->planar){int planes=out->planar?out->ch_count:1;for(ch=0;ch< planes;ch++){ctx->simd_f(out-> ch ch
Definition: audioconvert.c:56
The reader does not expect b to be semantically here and if the code is changed by maybe adding a a division or other the signedness will almost certainly be mistaken To avoid this confusion a new type was SUINT is the C unsigned type but it holds a signed int to use the same example SUINT a
Definition: undefined.txt:36
return FFERROR_NOT_READY
#define src
Definition: vp8dsp.c:254
EqParameter params[NBANDS+1]
#define N
Definition: af_mcompand.c:54
void * av_calloc(size_t nmemb, size_t size)
Non-inlined equivalent of av_mallocz_array().
Definition: mem.c:244
AVFilterFormats * ff_make_format_list(const int *fmts)
Create a list of supported formats.
Definition: formats.c:283
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
static float hn_imp(int n)
#define av_cold
Definition: attributes.h:82
#define fs(width, name, subs,...)
Definition: cbs_vp9.c:259
AVOptions.
static int equ_init(SuperEqualizerContext *s, int wb)
#define f(width, name)
Definition: cbs_vp9.c:255
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:388
static float hn(int n, EqParameter *param, float fs)
static int config_output(AVFilterLink *outlink)
#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)
Forward the status on an output link to an input link.
Definition: filters.h:199
A filter pad used for either input or output.
Definition: internal.h:54
static float hn_lpf(int n, float f, float fs)
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:259
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
static const AVFilterPad superequalizer_inputs[]
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 float izero(SuperEqualizerContext *s, float x)
GLenum GLint * params
Definition: opengl_enc.c:113
Definition: avfft.h:73
#define OFFSET(x)
void av_rdft_calc(RDFTContext *s, FFTSample *data)
int channels
number of audio channels, only used for audio.
Definition: frame.h:601
#define AF
AVFormatContext * ctx
Definition: movenc.c:48
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
#define s(width, name)
Definition: cbs_vp9.c:257
Definition: avfft.h:72
void av_rdft_end(RDFTContext *s)
int n
Definition: avisynth_c.h:760
RDFTContext * av_rdft_init(int nbits, enum RDFTransformType trans)
Set up a real FFT.
static const AVFilterPad outputs[]
Definition: af_acontrast.c:203
A list of supported channel layouts.
Definition: formats.h:85
AVFilter ff_af_superequalizer
#define sinf(x)
Definition: libm.h:419
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
int ff_inlink_consume_samples(AVFilterLink *link, unsigned min, unsigned max, AVFrame **rframe)
Take samples from the link&#39;s FIFO and update the link&#39;s stats.
Definition: avfilter.c:1500
FFT functions.
static const float bands[]
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
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
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
float im
Definition: fft.c:82
const char * name
Filter name.
Definition: avfilter.h:148
static void process_param(float *bc, EqParameter *param, float fs)
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
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_DEFINE_CLASS(superequalizer)
static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
FF_FILTER_FORWARD_WANTED(outlink, inlink)
A list of supported formats for one end of a filter link.
Definition: formats.h:64
static av_cold void uninit(AVFilterContext *ctx)
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 av_freep(p)
#define M_PI
Definition: mathematics.h:52
static int config_input(AVFilterLink *inlink)
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:410
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
#define M
static int activate(AVFilterContext *ctx)
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
static av_cold int init(AVFilterContext *ctx)
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:556