FFmpeg
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 
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:550
#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:295
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:1093
#define av_cold
Definition: attributes.h:82
#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:388
#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: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:569
if no frame is available
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:202
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:887
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
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
if(ret)
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
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 &&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
AVFilterFormats * ff_all_samplerates(void)
Definition: formats.c:394
static int request_frame(AVFilterLink *outlink)
Definition: af_anlmdn.c:309
#define flags(name, subs,...)
Definition: cbs_av1.c:561
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
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
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:155
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:701
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:1373
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
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:409
uint8_t ** extended_data
pointers to the data planes/channels.
Definition: frame.h:342
int nb_samples
number of audio samples (per channel) described by this frame
Definition: frame.h:361
for(j=16;j >0;--j)
int ff_set_common_samplerates(AVFilterContext *ctx, AVFilterFormats *samplerates)
Definition: formats.c:557
#define AV_NOPTS_VALUE
Undefined timestamp value.
Definition: avutil.h:248