FFmpeg  4.3
af_anlmdn.c
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1 /*
2  * Copyright (c) 2019 Paul B Mahol
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 <float.h>
22 
23 #include "libavutil/avassert.h"
24 #include "libavutil/audio_fifo.h"
25 #include "libavutil/avstring.h"
26 #include "libavutil/opt.h"
27 #include "avfilter.h"
28 #include "audio.h"
29 #include "formats.h"
30 
31 #include "af_anlmdndsp.h"
32 
33 #define WEIGHT_LUT_NBITS 20
34 #define WEIGHT_LUT_SIZE (1<<WEIGHT_LUT_NBITS)
35 
36 #define SQR(x) ((x) * (x))
37 
38 typedef struct AudioNLMeansContext {
39  const AVClass *class;
40 
41  float a;
42  int64_t pd;
43  int64_t rd;
44  float m;
45  int om;
46 
49 
50  int K;
51  int S;
52  int N;
53  int H;
54 
55  int offset;
58 
59  int64_t pts;
60 
62  int eof_left;
63 
66 
67 enum OutModes {
72 };
73 
74 #define OFFSET(x) offsetof(AudioNLMeansContext, x)
75 #define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
76 #define AFT AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
77 
78 static const AVOption anlmdn_options[] = {
79  { "s", "set denoising strength", OFFSET(a), AV_OPT_TYPE_FLOAT, {.dbl=0.00001},0.00001, 10, AFT },
80  { "p", "set patch duration", OFFSET(pd), AV_OPT_TYPE_DURATION, {.i64=2000}, 1000, 100000, AF },
81  { "r", "set research duration", OFFSET(rd), AV_OPT_TYPE_DURATION, {.i64=6000}, 2000, 300000, AF },
82  { "o", "set output mode", OFFSET(om), AV_OPT_TYPE_INT, {.i64=OUT_MODE}, 0, NB_MODES-1, AFT, "mode" },
83  { "i", "input", 0, AV_OPT_TYPE_CONST, {.i64=IN_MODE}, 0, 0, AFT, "mode" },
84  { "o", "output", 0, AV_OPT_TYPE_CONST, {.i64=OUT_MODE}, 0, 0, AFT, "mode" },
85  { "n", "noise", 0, AV_OPT_TYPE_CONST, {.i64=NOISE_MODE},0, 0, AFT, "mode" },
86  { "m", "set smooth factor", OFFSET(m), AV_OPT_TYPE_FLOAT, {.dbl=11.}, 1, 15, AF },
87  { NULL }
88 };
89 
90 AVFILTER_DEFINE_CLASS(anlmdn);
91 
93 {
96  static const enum AVSampleFormat sample_fmts[] = {
99  };
100  int ret;
101 
102  formats = ff_make_format_list(sample_fmts);
103  if (!formats)
104  return AVERROR(ENOMEM);
105  ret = ff_set_common_formats(ctx, formats);
106  if (ret < 0)
107  return ret;
108 
109  layouts = ff_all_channel_counts();
110  if (!layouts)
111  return AVERROR(ENOMEM);
112 
113  ret = ff_set_common_channel_layouts(ctx, layouts);
114  if (ret < 0)
115  return ret;
116 
117  formats = ff_all_samplerates();
118  return ff_set_common_samplerates(ctx, formats);
119 }
120 
121 static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
122 {
123  float distance = 0.;
124 
125  for (int k = -K; k <= K; k++)
126  distance += SQR(f1[k] - f2[k]);
127 
128  return distance;
129 }
130 
131 static void compute_cache_c(float *cache, const float *f,
132  ptrdiff_t S, ptrdiff_t K,
133  ptrdiff_t i, ptrdiff_t jj)
134 {
135  int v = 0;
136 
137  for (int j = jj; j < jj + S; j++, v++)
138  cache[v] += -SQR(f[i - K - 1] - f[j - K - 1]) + SQR(f[i + K] - f[j + K]);
139 }
140 
142 {
145 
146  if (ARCH_X86)
147  ff_anlmdn_init_x86(dsp);
148 }
149 
150 static int config_output(AVFilterLink *outlink)
151 {
152  AVFilterContext *ctx = outlink->src;
153  AudioNLMeansContext *s = ctx->priv;
154  int ret;
155 
156  s->K = av_rescale(s->pd, outlink->sample_rate, AV_TIME_BASE);
157  s->S = av_rescale(s->rd, outlink->sample_rate, AV_TIME_BASE);
158 
159  s->eof_left = -1;
160  s->pts = AV_NOPTS_VALUE;
161  s->H = s->K * 2 + 1;
162  s->N = s->H + (s->K + s->S) * 2;
163 
164  av_log(ctx, AV_LOG_DEBUG, "K:%d S:%d H:%d N:%d\n", s->K, s->S, s->H, s->N);
165 
166  av_frame_free(&s->in);
167  av_frame_free(&s->cache);
168  s->in = ff_get_audio_buffer(outlink, s->N);
169  if (!s->in)
170  return AVERROR(ENOMEM);
171 
172  s->cache = ff_get_audio_buffer(outlink, s->S * 2);
173  if (!s->cache)
174  return AVERROR(ENOMEM);
175 
176  s->fifo = av_audio_fifo_alloc(outlink->format, outlink->channels, s->N);
177  if (!s->fifo)
178  return AVERROR(ENOMEM);
179 
180  ret = av_audio_fifo_write(s->fifo, (void **)s->in->extended_data, s->K + s->S);
181  if (ret < 0)
182  return ret;
183 
184  s->pdiff_lut_scale = 1.f / s->m * WEIGHT_LUT_SIZE;
185  for (int i = 0; i < WEIGHT_LUT_SIZE; i++) {
186  float w = -i / s->pdiff_lut_scale;
187 
188  s->weight_lut[i] = expf(w);
189  }
190 
191  ff_anlmdn_init(&s->dsp);
192 
193  return 0;
194 }
195 
196 static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
197 {
198  AudioNLMeansContext *s = ctx->priv;
199  AVFrame *out = arg;
200  const int S = s->S;
201  const int K = s->K;
202  const int om = s->om;
203  const float *f = (const float *)(s->in->extended_data[ch]) + K;
204  float *cache = (float *)s->cache->extended_data[ch];
205  const float sw = (65536.f / (4 * K + 2)) / sqrtf(s->a);
206  float *dst = (float *)out->extended_data[ch] + s->offset;
207  const float smooth = s->m;
208 
209  for (int i = S; i < s->H + S; i++) {
210  float P = 0.f, Q = 0.f;
211  int v = 0;
212 
213  if (i == S) {
214  for (int j = i - S; j <= i + S; j++) {
215  if (i == j)
216  continue;
217  cache[v++] = s->dsp.compute_distance_ssd(f + i, f + j, K);
218  }
219  } else {
220  s->dsp.compute_cache(cache, f, S, K, i, i - S);
221  s->dsp.compute_cache(cache + S, f, S, K, i, i + 1);
222  }
223 
224  for (int j = 0; j < 2 * S && !ctx->is_disabled; j++) {
225  const float distance = cache[j];
226  unsigned weight_lut_idx;
227  float w;
228 
229  if (distance < 0.f) {
230  cache[j] = 0.f;
231  continue;
232  }
233  w = distance * sw;
234  if (w >= smooth)
235  continue;
236  weight_lut_idx = w * s->pdiff_lut_scale;
237  av_assert2(weight_lut_idx < WEIGHT_LUT_SIZE);
238  w = s->weight_lut[weight_lut_idx];
239  P += w * f[i - S + j + (j >= S)];
240  Q += w;
241  }
242 
243  P += f[i];
244  Q += 1;
245 
246  switch (om) {
247  case IN_MODE: dst[i - S] = f[i]; break;
248  case OUT_MODE: dst[i - S] = P / Q; break;
249  case NOISE_MODE: dst[i - S] = f[i] - (P / Q); break;
250  }
251  }
252 
253  return 0;
254 }
255 
256 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
257 {
258  AVFilterContext *ctx = inlink->dst;
259  AVFilterLink *outlink = ctx->outputs[0];
260  AudioNLMeansContext *s = ctx->priv;
261  AVFrame *out = NULL;
262  int available, wanted, ret;
263 
264  if (s->pts == AV_NOPTS_VALUE)
265  s->pts = in->pts;
266 
267  ret = av_audio_fifo_write(s->fifo, (void **)in->extended_data,
268  in->nb_samples);
269  av_frame_free(&in);
270 
271  s->offset = 0;
272  available = av_audio_fifo_size(s->fifo);
273  wanted = (available / s->H) * s->H;
274 
275  if (wanted >= s->H && available >= s->N) {
276  out = ff_get_audio_buffer(outlink, wanted);
277  if (!out)
278  return AVERROR(ENOMEM);
279  }
280 
281  while (available >= s->N) {
282  ret = av_audio_fifo_peek(s->fifo, (void **)s->in->extended_data, s->N);
283  if (ret < 0)
284  break;
285 
286  ctx->internal->execute(ctx, filter_channel, out, NULL, inlink->channels);
287 
288  av_audio_fifo_drain(s->fifo, s->H);
289 
290  s->offset += s->H;
291  available -= s->H;
292  }
293 
294  if (out) {
295  out->pts = s->pts;
296  out->nb_samples = s->offset;
297  if (s->eof_left >= 0) {
298  out->nb_samples = FFMIN(s->eof_left, s->offset);
299  s->eof_left -= out->nb_samples;
300  }
301  s->pts += av_rescale_q(s->offset, (AVRational){1, outlink->sample_rate}, outlink->time_base);
302 
303  return ff_filter_frame(outlink, out);
304  }
305 
306  return ret;
307 }
308 
309 static int request_frame(AVFilterLink *outlink)
310 {
311  AVFilterContext *ctx = outlink->src;
312  AudioNLMeansContext *s = ctx->priv;
313  int ret;
314 
315  ret = ff_request_frame(ctx->inputs[0]);
316 
317  if (ret == AVERROR_EOF && s->eof_left != 0) {
318  AVFrame *in;
319 
320  if (s->eof_left < 0)
321  s->eof_left = av_audio_fifo_size(s->fifo) - (s->S + s->K);
322  if (s->eof_left <= 0)
323  return AVERROR_EOF;
324  in = ff_get_audio_buffer(outlink, s->H);
325  if (!in)
326  return AVERROR(ENOMEM);
327 
328  return filter_frame(ctx->inputs[0], in);
329  }
330 
331  return ret;
332 }
333 
335 {
336  AudioNLMeansContext *s = ctx->priv;
337 
339  av_frame_free(&s->in);
340  av_frame_free(&s->cache);
341 }
342 
343 static const AVFilterPad inputs[] = {
344  {
345  .name = "default",
346  .type = AVMEDIA_TYPE_AUDIO,
347  .filter_frame = filter_frame,
348  },
349  { NULL }
350 };
351 
352 static const AVFilterPad outputs[] = {
353  {
354  .name = "default",
355  .type = AVMEDIA_TYPE_AUDIO,
356  .config_props = config_output,
357  .request_frame = request_frame,
358  },
359  { NULL }
360 };
361 
363  .name = "anlmdn",
364  .description = NULL_IF_CONFIG_SMALL("Reduce broadband noise from stream using Non-Local Means."),
365  .query_formats = query_formats,
366  .priv_size = sizeof(AudioNLMeansContext),
367  .priv_class = &anlmdn_class,
368  .uninit = uninit,
369  .inputs = inputs,
370  .outputs = outputs,
374 };
static const AVOption anlmdn_options[]
Definition: af_anlmdn.c:78
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:581
#define P
AVAudioFifo * av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels, int nb_samples)
Allocate an AVAudioFifo.
Definition: audio_fifo.c:59
This structure describes decoded (raw) audio or video data.
Definition: frame.h:300
if(ret< 0)
Definition: vf_mcdeint.c:279
AVOption.
Definition: opt.h:246
static float compute_distance_ssd_c(const float *f1, const float *f2, ptrdiff_t K)
Definition: af_anlmdn.c:121
Main libavfilter public API header.
void av_audio_fifo_free(AVAudioFifo *af)
Free an AVAudioFifo.
Definition: audio_fifo.c:45
#define AFT
Definition: af_anlmdn.c:76
#define SQR(x)
Definition: af_anlmdn.c:36
void ff_anlmdn_init(AudioNLMDNDSPContext *dsp)
Definition: af_anlmdn.c:141
int is_disabled
the enabled state from the last expression evaluation
Definition: avfilter.h:385
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:1075
#define av_cold
Definition: attributes.h:88
#define av_assert2(cond)
assert() equivalent, that does lie in speed critical code.
Definition: avassert.h:64
AVOptions.
#define f(width, name)
Definition: cbs_vp9.c:255
AVFilter ff_af_anlmdn
Definition: af_anlmdn.c:362
int64_t pts
Presentation timestamp in time_base units (time when frame should be shown to user).
Definition: frame.h:393
#define AVERROR_EOF
End of file.
Definition: error.h:55
static int config_output(AVFilterLink *outlink)
Definition: af_anlmdn.c:150
float(* compute_distance_ssd)(const float *f1, const float *f2, ptrdiff_t K)
Definition: af_anlmdndsp.h:32
#define av_log(a,...)
A filter pad used for either input or output.
Definition: internal.h:54
int64_t av_rescale_q(int64_t a, AVRational bq, AVRational cq)
Rescale a 64-bit integer by 2 rational numbers.
Definition: mathematics.c:142
#define expf(x)
Definition: libm.h:283
#define i(width, name, range_min, range_max)
Definition: cbs_h2645.c:269
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:600
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 ARCH_X86
Definition: config.h:38
#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:203
void ff_anlmdn_init_x86(AudioNLMDNDSPContext *s)
#define NULL_IF_CONFIG_SMALL(x)
Return NULL if CONFIG_SMALL is true, otherwise the argument without modification. ...
Definition: internal.h:186
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:869
void * priv
private data for use by the filter
Definition: avfilter.h:353
#define AVFILTER_FLAG_SLICE_THREADS
The filter supports multithreading by splitting frames into multiple parts and processing them concur...
Definition: avfilter.h:116
#define AV_LOG_DEBUG
Stuff which is only useful for libav* developers.
Definition: log.h:197
const char * arg
Definition: jacosubdec.c:66
simple assert() macros that are a bit more flexible than ISO C assert().
static const AVFilterPad outputs[]
Definition: af_anlmdn.c:352
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/(UINT64_C(1)<< 63))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double, AV_SAMPLE_FMT_S64, *(const int64_t *) pi *(1.0/(UINT64_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 *(UINT64_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 *(UINT64_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 &&HAVE_MMX) 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
Context for an Audio FIFO Buffer.
Definition: audio_fifo.c:34
static float distance(float x, float y, int band)
int av_audio_fifo_size(AVAudioFifo *af)
Get the current number of samples in the AVAudioFifo available for reading.
Definition: audio_fifo.c:228
int64_t av_rescale(int64_t a, int64_t b, int64_t c)
Rescale a 64-bit integer with rounding to nearest.
Definition: mathematics.c:129
#define AV_TIME_BASE
Internal time base represented as integer.
Definition: avutil.h:254
#define FFMIN(a, b)
Definition: common.h:96
uint8_t w
Definition: llviddspenc.c:38
float weight_lut[WEIGHT_LUT_SIZE]
Definition: af_anlmdn.c:48
AVFormatContext * ctx
Definition: movenc.c:48
static int query_formats(AVFilterContext *ctx)
Definition: af_anlmdn.c:92
#define s(width, name)
Definition: cbs_vp9.c:257
OutModes
Definition: af_afftdn.c:37
static int filter_channel(AVFilterContext *ctx, void *arg, int ch, int nb_jobs)
Definition: af_anlmdn.c:196
#define AF
Definition: af_anlmdn.c:75
static const AVFilterPad inputs[]
Definition: af_anlmdn.c:343
A list of supported channel layouts.
Definition: formats.h:85
AVFILTER_DEFINE_CLASS(anlmdn)
void(* compute_cache)(float *cache, const float *f, ptrdiff_t S, ptrdiff_t K, ptrdiff_t i, ptrdiff_t jj)
Definition: af_anlmdndsp.h:33
AVSampleFormat
Audio sample formats.
Definition: samplefmt.h:58
static void compute_cache_c(float *cache, const float *f, ptrdiff_t S, ptrdiff_t K, ptrdiff_t i, ptrdiff_t jj)
Definition: af_anlmdn.c:131
static float smooth(DeshakeOpenCLContext *deshake_ctx, float *gauss_kernel, int length, float max_val, AVFifoBuffer *values)
Describe the class of an AVClass context structure.
Definition: log.h:67
Filter definition.
Definition: avfilter.h:144
Rational number (pair of numerator and denominator).
Definition: rational.h:58
const char * name
Filter name.
Definition: avfilter.h:148
#define AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL
Same as AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC, except that the filter will have its filter_frame() c...
Definition: avfilter.h:133
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:425
static int request_frame(AVFilterLink *outlink)
Definition: af_anlmdn.c:309
#define flags(name, subs,...)
Definition: cbs_av1.c:564
AVFilterInternal * internal
An opaque struct for libavfilter internal use.
Definition: avfilter.h:378
int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
Write data to an AVAudioFifo.
Definition: audio_fifo.c:112
#define WEIGHT_LUT_SIZE
Definition: af_anlmdn.c:34
AVFrame * cache
Definition: af_anlmdn.c:57
int av_audio_fifo_drain(AVAudioFifo *af, int nb_samples)
Drain data from an AVAudioFifo.
Definition: audio_fifo.c:201
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
Definition: af_anlmdn.c:256
static av_cold void uninit(AVFilterContext *ctx)
Definition: af_anlmdn.c:334
avfilter_execute_func * execute
Definition: internal.h:144
Audio FIFO Buffer.
#define OFFSET(x)
Definition: af_anlmdn.c:74
A list of supported formats for one end of a filter link.
Definition: formats.h:64
AVAudioFifo * fifo
Definition: af_anlmdn.c:61
AudioNLMDNDSPContext dsp
Definition: af_anlmdn.c:64
int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
Peek data from an AVAudioFifo.
Definition: audio_fifo.c:138
An instance of a filter.
Definition: avfilter.h:338
static enum AVSampleFormat sample_fmts[]
Definition: adpcmenc.c:731
FILE * out
Definition: movenc.c:54
static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)
Definition: af_afftdn.c:1374
int ff_request_frame(AVFilterLink *link)
Request an input frame from the filter at the other end of the link.
Definition: avfilter.c:407
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:440
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:347
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:366
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:588
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248