FFmpeg: libavfilter/af_adelay.c Source File

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af_adelay.c

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1 /*

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 */

21 #include "libavutil/avstring.h"

22 #include "libavutil/eval.h"

23 #include "libavutil/opt.h"

24 #include "libavutil/samplefmt.h"

25 #include "avfilter.h"

26 #include "audio.h"

27 #include "filters.h"

28 #include "internal.h"

30 typedef struct ChanDelay {

31 int64_t delay;

32 size_t delay_index;

33 size_t index;

34 unsigned int samples_size;

35 uint8_t *samples;

36 } ChanDelay;

38 typedef struct AudioDelayContext {

39 const AVClass *class;

40 int all;

41 char *delays;

42 ChanDelay *chandelay;

43 int nb_delays;

44 int block_align;

45 int64_t padding;

46 int64_t max_delay;

47 int64_t offset;

48 int64_t next_pts;

49 int eof;

51 AVFrame *input;

53 void (*delay_channel)(ChanDelay *d, int nb_samples,

54 const uint8_t *src, uint8_t *dst);

55 int (*resize_channel_samples)(ChanDelay *d, int64_t new_delay);

56 } AudioDelayContext;

58 #define OFFSET(x) offsetof(AudioDelayContext, x)

59 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

61 static const AVOption adelay_options[] = {

62 { "delays", "set list of delays for each channel", OFFSET(delays), AV_OPT_TYPE_STRING, {.str=NULL}, 0, 0, A | AV_OPT_FLAG_RUNTIME_PARAM },

63 { "all", "use last available delay for remained channels", OFFSET(all), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },

64 { NULL }

65 };

67 AVFILTER_DEFINE_CLASS(adelay);

69 #define DELAY(name, type, fill) \

70 static void delay_channel_## name ##p(ChanDelay *d, int nb_samples, \

71 const uint8_t *ssrc, uint8_t *ddst) \

72 { \

73 const type *src = (type *)ssrc; \

74 type *dst = (type *)ddst; \

75 type *samples = (type *)d->samples; \

76 \

77 while (nb_samples) { \

78 if (d->delay_index < d->delay) { \

79 const int len = FFMIN(nb_samples, d->delay - d->delay_index); \

80 \

81 memcpy(&samples[d->delay_index], src, len * sizeof(type)); \

82 memset(dst, fill, len * sizeof(type)); \

83 d->delay_index += len; \

84 src += len; \

85 dst += len; \

86 nb_samples -= len; \

87 } else { \

88 *dst = samples[d->index]; \

89 samples[d->index] = *src; \

90 nb_samples--; \

91 d->index++; \

92 src++, dst++; \

93 d->index = d->index >= d->delay ? 0 : d->index; \

94 } \

95 } \

96 }

98 DELAY(u8, uint8_t, 0x80)

99 DELAY(s16, int16_t, 0)

100 DELAY(s32, int32_t, 0)

101 DELAY(flt, float, 0)

102 DELAY(dbl, double, 0)

103

104 #define CHANGE_DELAY(name, type, fill) \

105 static int resize_samples_## name ##p(ChanDelay *d, int64_t new_delay) \

106 { \

107 type *samples; \

108 \

109 if (new_delay == d->delay) { \

110 return 0; \

111 } \

112 \

113 if (new_delay == 0) { \

114 av_freep(&d->samples); \

115 d->samples_size = 0; \

116 d->delay = 0; \

117 d->index = 0; \

118 d->delay_index = 0; \

119 return 0; \

120 } \

121 \

122 samples = (type *) av_fast_realloc(d->samples, &d->samples_size, new_delay * sizeof(type)); \

123 if (!samples) { \

124 return AVERROR(ENOMEM); \

125 } \

126 \

127 if (new_delay < d->delay) { \

128 if (d->index > new_delay) { \

129 d->index -= new_delay; \

130 memmove(samples, &samples[new_delay], d->index * sizeof(type)); \

131 d->delay_index = new_delay; \

132 } else if (d->delay_index > d->index) { \

133 memmove(&samples[d->index], &samples[d->index+(d->delay-new_delay)], \

134 (new_delay - d->index) * sizeof(type)); \

135 d->delay_index -= d->delay - new_delay; \

136 } \

137 } else { \

138 size_t block_size; \

139 if (d->delay_index >= d->delay) { \

140 block_size = (d->delay - d->index) * sizeof(type); \

141 memmove(&samples[d->index+(new_delay - d->delay)], &samples[d->index], block_size); \

142 d->delay_index = new_delay; \

143 } else { \

144 d->delay_index += new_delay - d->delay; \

145 } \

146 block_size = (new_delay - d->delay) * sizeof(type); \

147 memset(&samples[d->index], fill, block_size); \

148 } \

149 d->delay = new_delay; \

150 d->samples = (void *) samples; \

151 return 0; \

152 }

153

154 CHANGE_DELAY(u8, uint8_t, 0x80)

155 CHANGE_DELAY(s16, int16_t, 0)

156 CHANGE_DELAY(s32, int32_t, 0)

157 CHANGE_DELAY(flt, float, 0)

158 CHANGE_DELAY(dbl, double, 0)

159

160 static int parse_delays(char *p, char **saveptr, int64_t *result, AVFilterContext *ctx, int sample_rate) {

161 float delay, div;

162 int ret;

163 char *arg;

164 char type = 0;

165

166 if (!(arg = av_strtok(p, "|", saveptr)))

167 return 1;

168

169 ret = av_sscanf(arg, "%"SCNd64"%c", result, &type);

170 if (ret != 2 || type != 'S') {

171 div = type == 's' ? 1.0 : 1000.0;

172 if (av_sscanf(arg, "%f", &delay) != 1) {

173 av_log(ctx, AV_LOG_ERROR, "Invalid syntax for delay.\n");

174 return AVERROR(EINVAL);

175 }

176 *result = delay * sample_rate / div;

177 }

178

179 if (*result < 0) {

180 av_log(ctx, AV_LOG_ERROR, "Delay must be non negative number.\n");

181 return AVERROR(EINVAL);

182 }

183 return 0;

184 }

185

186 static int config_input(AVFilterLink *inlink)

187 {

188 AVFilterContext *ctx = inlink->dst;

189 AudioDelayContext *s = ctx->priv;

190 char *p, *saveptr = NULL;

191 int i;

192

193 s->next_pts = AV_NOPTS_VALUE;

194 s->chandelay = av_calloc(inlink->ch_layout.nb_channels, sizeof(*s->chandelay));

195 if (!s->chandelay)

196 return AVERROR(ENOMEM);

197 s->nb_delays = inlink->ch_layout.nb_channels;

198 s->block_align = av_get_bytes_per_sample(inlink->format);

199

200 p = s->delays;

201 for (i = 0; i < s->nb_delays; i++) {

202 ChanDelay *d = &s->chandelay[i];

203 int ret;

204

205 ret = parse_delays(p, &saveptr, &d->delay, ctx, inlink->sample_rate);

206 if (ret == 1)

207 break;

208 else if (ret < 0)

209 return ret;

210 p = NULL;

211 }

212

213 if (s->all && i) {

214 for (int j = i; j < s->nb_delays; j++)

215 s->chandelay[j].delay = s->chandelay[i-1].delay;

216 }

217

218 s->padding = s->chandelay[0].delay;

219 for (i = 1; i < s->nb_delays; i++) {

220 ChanDelay *d = &s->chandelay[i];

221

222 s->padding = FFMIN(s->padding, d->delay);

223 }

224

225 if (s->padding) {

226 for (i = 0; i < s->nb_delays; i++) {

227 ChanDelay *d = &s->chandelay[i];

228

229 d->delay -= s->padding;

230 }

231

232 s->offset = av_rescale_q(s->padding,

233 av_make_q(1, inlink->sample_rate),

234 inlink->time_base);

235 }

236

237 for (i = 0; i < s->nb_delays; i++) {

238 ChanDelay *d = &s->chandelay[i];

239

240 if (!d->delay)

241 continue;

242

243 if (d->delay > SIZE_MAX) {

244 av_log(ctx, AV_LOG_ERROR, "Requested delay is too big.\n");

245 return AVERROR(EINVAL);

246 }

247

248 d->samples = av_malloc_array(d->delay, s->block_align);

249 if (!d->samples)

250 return AVERROR(ENOMEM);

251 d->samples_size = d->delay * s->block_align;

252

253 s->max_delay = FFMAX(s->max_delay, d->delay);

254 }

255

256 switch (inlink->format) {

257 case AV_SAMPLE_FMT_U8P : s->delay_channel = delay_channel_u8p ;

258 s->resize_channel_samples = resize_samples_u8p; break;

259 case AV_SAMPLE_FMT_S16P: s->delay_channel = delay_channel_s16p;

260 s->resize_channel_samples = resize_samples_s16p; break;

261 case AV_SAMPLE_FMT_S32P: s->delay_channel = delay_channel_s32p;

262 s->resize_channel_samples = resize_samples_s32p; break;

263 case AV_SAMPLE_FMT_FLTP: s->delay_channel = delay_channel_fltp;

264 s->resize_channel_samples = resize_samples_fltp; break;

265 case AV_SAMPLE_FMT_DBLP: s->delay_channel = delay_channel_dblp;

266 s->resize_channel_samples = resize_samples_dblp; break;

267 }

268

269 return 0;

270 }

271

272 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,

273 char *res, int res_len, int flags)

274 {

275 int ret = AVERROR(ENOSYS);

276 AVFilterLink *inlink = ctx->inputs[0];

277 AudioDelayContext *s = ctx->priv;

278

279 if (!strcmp(cmd, "delays")) {

280 int64_t delay;

281 char *p, *saveptr = NULL;

282 int64_t all_delay = -1;

283 int64_t max_delay = 0;

284 char *args_cpy = av_strdup(args);

285 if (args_cpy == NULL) {

286 return AVERROR(ENOMEM);

287 }

288

289 ret = 0;

290 p = args_cpy;

291

292 if (!strncmp(args, "all:", 4)) {

293 p = &args_cpy[4];

294 ret = parse_delays(p, &saveptr, &all_delay, ctx, inlink->sample_rate);

295 if (ret == 1)

296 ret = AVERROR(EINVAL);

297 else if (ret == 0)

298 delay = all_delay;

299 }

300

301 if (!ret) {

302 for (int i = 0; i < s->nb_delays; i++) {

303 ChanDelay *d = &s->chandelay[i];

304

305 if (all_delay < 0) {

306 ret = parse_delays(p, &saveptr, &delay, ctx, inlink->sample_rate);

307 if (ret != 0) {

308 ret = 0;

309 break;

310 }

311 p = NULL;

312 }

313

314 ret = s->resize_channel_samples(d, delay);

315 if (ret)

316 break;

317 max_delay = FFMAX(max_delay, d->delay);

318 }

319 s->max_delay = FFMAX(s->max_delay, max_delay);

320 }

321 av_freep(&args_cpy);

322 }

323 return ret;

324 }

325

326 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)

327 {

328 AVFilterContext *ctx = inlink->dst;

329 AVFilterLink *outlink = ctx->outputs[0];

330 AudioDelayContext *s = ctx->priv;

331 AVFrame *out_frame;

332 int i;

333

334 if (ctx->is_disabled || !s->delays) {

335 s->input = NULL;

336 return ff_filter_frame(outlink, frame);

337 }

338

339 s->next_pts = av_rescale_q(frame->pts, inlink->time_base, outlink->time_base);

340

341 out_frame = ff_get_audio_buffer(outlink, frame->nb_samples);

342 if (!out_frame) {

343 s->input = NULL;

344 av_frame_free(&frame);

345 return AVERROR(ENOMEM);

346 }

347 av_frame_copy_props(out_frame, frame);

348

349 for (i = 0; i < s->nb_delays; i++) {

350 ChanDelay *d = &s->chandelay[i];

351 const uint8_t *src = frame->extended_data[i];

352 uint8_t *dst = out_frame->extended_data[i];

353

354 if (!d->delay)

355 memcpy(dst, src, frame->nb_samples * s->block_align);

356 else

357 s->delay_channel(d, frame->nb_samples, src, dst);

358 }

359

360 out_frame->pts = s->next_pts + s->offset;

361 out_frame->duration = av_rescale_q(out_frame->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);

362 s->next_pts += out_frame->duration;

363 av_frame_free(&frame);

364 s->input = NULL;

365 return ff_filter_frame(outlink, out_frame);

366 }

367

368 static int activate(AVFilterContext *ctx)

369 {

370 AVFilterLink *inlink = ctx->inputs[0];

371 AVFilterLink *outlink = ctx->outputs[0];

372 AudioDelayContext *s = ctx->priv;

373 AVFrame *frame = NULL;

374 int ret, status;

375 int64_t pts;

376

377 FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);

378

379 if (!s->input) {

380 ret = ff_inlink_consume_frame(inlink, &s->input);

381 if (ret < 0)

382 return ret;

383 }

384

385 if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {

386 if (status == AVERROR_EOF)

387 s->eof = 1;

388 }

389

390 if (s->next_pts == AV_NOPTS_VALUE && pts != AV_NOPTS_VALUE)

391 s->next_pts = av_rescale_q(pts, inlink->time_base, outlink->time_base);

392

393 if (s->padding) {

394 int nb_samples = FFMIN(s->padding, 2048);

395

396 frame = ff_get_audio_buffer(outlink, nb_samples);

397 if (!frame)

398 return AVERROR(ENOMEM);

399 s->padding -= nb_samples;

400

401 av_samples_set_silence(frame->extended_data, 0,

402 frame->nb_samples,

403 outlink->ch_layout.nb_channels,

404 frame->format);

405

406 frame->duration = av_rescale_q(frame->nb_samples,

407 (AVRational){1, outlink->sample_rate},

408 outlink->time_base);

409 frame->pts = s->next_pts;

410 s->next_pts += frame->duration;

411

412 return ff_filter_frame(outlink, frame);

413 }

414

415 if (s->input)

416 return filter_frame(inlink, s->input);

417

418 if (s->eof && s->max_delay) {

419 int nb_samples = FFMIN(s->max_delay, 2048);

420

421 frame = ff_get_audio_buffer(outlink, nb_samples);

422 if (!frame)

423 return AVERROR(ENOMEM);

424 s->max_delay -= nb_samples;

425

426 av_samples_set_silence(frame->extended_data, 0,

427 frame->nb_samples,

428 outlink->ch_layout.nb_channels,

429 frame->format);

430

431 frame->duration = av_rescale_q(frame->nb_samples,

432 (AVRational){1, outlink->sample_rate},

433 outlink->time_base);

434 frame->pts = s->next_pts;

435 s->next_pts += frame->duration;

436 return filter_frame(inlink, frame);

437 }

438

439 if (s->eof && s->max_delay == 0) {

440 ff_outlink_set_status(outlink, AVERROR_EOF, s->next_pts);

441 return 0;

442 }

443

444 if (!s->eof)

445 FF_FILTER_FORWARD_WANTED(outlink, inlink);

446

447 return FFERROR_NOT_READY;

448 }

449

450 static av_cold void uninit(AVFilterContext *ctx)

451 {

452 AudioDelayContext *s = ctx->priv;

453

454 if (s->chandelay) {

455 for (int i = 0; i < s->nb_delays; i++)

456 av_freep(&s->chandelay[i].samples);

457 }

458 av_freep(&s->chandelay);

459 }

460

461 static const AVFilterPad adelay_inputs[] = {

462 {

463 .name = "default",

464 .type = AVMEDIA_TYPE_AUDIO,

465 .config_props = config_input,

466 },

467 };

468

469 const AVFilter ff_af_adelay = {

470 .name = "adelay",

471 .description = NULL_IF_CONFIG_SMALL("Delay one or more audio channels."),

472 .priv_size = sizeof(AudioDelayContext),

473 .priv_class = &adelay_class,

474 .activate = activate,

475 .uninit = uninit,

476 FILTER_INPUTS(adelay_inputs),

477 FILTER_OUTPUTS(ff_audio_default_filterpad),

478 FILTER_SAMPLEFMTS(AV_SAMPLE_FMT_U8P, AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,

479 AV_SAMPLE_FMT_FLTP, AV_SAMPLE_FMT_DBLP),

480 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,

481 .process_command = process_command,

482 };

AudioDelayContext::next_pts

int64_t next_pts

Definition: af_adelay.c:48

ff_get_audio_buffer

AVFrame * ff_get_audio_buffer(AVFilterLink *link, int nb_samples)

Request an audio samples buffer with a specific set of permissions.

Definition: audio.c:97

AV_SAMPLE_FMT_FLTP

@ AV_SAMPLE_FMT_FLTP

float, planar

Definition: samplefmt.h:66

AudioDelayContext::resize_channel_samples

int(* resize_channel_samples)(ChanDelay *d, int64_t new_delay)

Definition: af_adelay.c:55

AVERROR

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

opt.h

ff_filter_frame

int ff_filter_frame(AVFilterLink *link, AVFrame *frame)

Send a frame of data to the next filter.

Definition: avfilter.c:1018

AVFrame::duration

int64_t duration

Duration of the frame, in the same units as pts.

Definition: frame.h:750

AVERROR_EOF

#define AVERROR_EOF

End of file.

Definition: error.h:57

FFERROR_NOT_READY

return FFERROR_NOT_READY

Definition: filter_design.txt:204

int64_t

long long int64_t

Definition: coverity.c:34

inlink

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

Definition: filter_design.txt:212

av_frame_free

void av_frame_free(AVFrame **frame)

Free the frame and any dynamically allocated objects in it, e.g.

Definition: frame.c:130

AVFrame

This structure describes decoded (raw) audio or video data.

Definition: frame.h:344

AVFrame::pts

int64_t pts

Presentation timestamp in time_base units (time when frame should be shown to user).

Definition: frame.h:456

av_samples_set_silence

int av_samples_set_silence(uint8_t *const *audio_data, int offset, int nb_samples, int nb_channels, enum AVSampleFormat sample_fmt)

Fill an audio buffer with silence.

Definition: samplefmt.c:246

AVOption

AVOption.

Definition: opt.h:346

AVFILTER_DEFINE_CLASS

AVFILTER_DEFINE_CLASS(adelay)

AV_SAMPLE_FMT_S32P

@ AV_SAMPLE_FMT_S32P

signed 32 bits, planar

Definition: samplefmt.h:65

ChanDelay

Definition: af_adelay.c:30

FFMAX

#define FFMAX(a, b)

Definition: macros.h:47

AVFilter::name

const char * name

Filter name.

Definition: avfilter.h:170

AVChannelLayout::nb_channels

int nb_channels

Number of channels in this layout.

Definition: channel_layout.h:313

ChanDelay::delay

int64_t delay

Definition: af_adelay.c:31

AVFilterLink

A link between two filters.

Definition: avfilter.h:542

FF_FILTER_FORWARD_STATUS_BACK

#define FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink)

Forward the status on an output link to an input link.

Definition: filters.h:199

ff_af_adelay

const AVFilter ff_af_adelay

Definition: af_adelay.c:469

sample_rate

Definition: ffmpeg_filter.c:409

OFFSET

#define OFFSET(x)

Definition: af_adelay.c:58

ff_inlink_consume_frame

int ff_inlink_consume_frame(AVFilterLink *link, AVFrame **rframe)

Take a frame from the link's FIFO and update the link's stats.

Definition: avfilter.c:1445

AudioDelayContext

Definition: af_adelay.c:38

samplefmt.h

type

it s the only field you need to keep assuming you have a context There is some magic you don t need to care about around this just let it vf type

Definition: writing_filters.txt:86

pts

static int64_t pts

Definition: transcode_aac.c:643

AVFilterPad

A filter pad used for either input or output.

Definition: internal.h:33

activate

static int activate(AVFilterContext *ctx)

Definition: af_adelay.c:368

config_input

static int config_input(AVFilterLink *inlink)

Definition: af_adelay.c:186

AV_LOG_ERROR

#define AV_LOG_ERROR

Something went wrong and cannot losslessly be recovered.

Definition: log.h:180

AudioDelayContext::delay_channel

void(* delay_channel)(ChanDelay *d, int nb_samples, const uint8_t *src, uint8_t *dst)

Definition: af_adelay.c:53

av_cold

#define av_cold

Definition: attributes.h:90

ff_outlink_set_status

static void ff_outlink_set_status(AVFilterLink *link, int status, int64_t pts)

Set the status field of a link from the source filter.

Definition: filters.h:189

#define s(width, name)

Definition: cbs_vp9.c:198

AudioDelayContext::delays

char * delays

Definition: af_adelay.c:41

AVMEDIA_TYPE_AUDIO

@ AVMEDIA_TYPE_AUDIO

Definition: avutil.h:202

av_strtok

char * av_strtok(char *s, const char *delim, char **saveptr)

Split the string into several tokens which can be accessed by successive calls to av_strtok().

Definition: avstring.c:178

filters.h

adelay_options

static const AVOption adelay_options[]

Definition: af_adelay.c:61

ctx

AVFormatContext * ctx

Definition: movenc.c:48

AudioDelayContext::nb_delays

int nb_delays

Definition: af_adelay.c:43

AudioDelayContext::padding

int64_t padding

Definition: af_adelay.c:45

av_rescale_q

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

FILTER_INPUTS

#define FILTER_INPUTS(array)

Definition: internal.h:182

frame

static AVFrame * frame

Definition: demux_decode.c:54

arg

const char * arg

Definition: jacosubdec.c:67

av_sscanf

int av_sscanf(const char *string, const char *format,...)

See libc sscanf manual for more information.

Definition: avsscanf.c:962

AVClass

Describe the class of an AVClass context structure.

Definition: log.h:66

result

and forward the result(frame or status change) to the corresponding input. If nothing is possible

NULL

#define NULL

Definition: coverity.c:32

av_frame_copy_props

int av_frame_copy_props(AVFrame *dst, const AVFrame *src)

Copy only "metadata" fields from src to dst.

Definition: frame.c:679

AVRational

Rational number (pair of numerator and denominator).

Definition: rational.h:58

parse_delays

static int parse_delays(char *p, char **saveptr, int64_t *result, AVFilterContext *ctx, int sample_rate)

Definition: af_adelay.c:160

#define A

Definition: af_adelay.c:59

ff_audio_default_filterpad

const AVFilterPad ff_audio_default_filterpad[1]

An AVFilterPad array whose only entry has name "default" and is of type AVMEDIA_TYPE_AUDIO.

Definition: audio.c:33

ff_inlink_acknowledge_status

int ff_inlink_acknowledge_status(AVFilterLink *link, int *rstatus, int64_t *rpts)

Test and acknowledge the change of status on the link.

Definition: avfilter.c:1392

eval.h

NULL_IF_CONFIG_SMALL

#define NULL_IF_CONFIG_SMALL(x)

Return NULL if CONFIG_SMALL is true, otherwise the argument without modification.

Definition: internal.h:106

AV_SAMPLE_FMT_U8P

@ AV_SAMPLE_FMT_U8P

unsigned 8 bits, planar

Definition: samplefmt.h:63

DELAY

#define DELAY(name, type, fill)

Definition: af_adelay.c:69

AudioDelayContext::block_align

int block_align

Definition: af_adelay.c:44

av_make_q

static AVRational av_make_q(int num, int den)

Create an AVRational.

Definition: rational.h:71

AV_NOPTS_VALUE

#define AV_NOPTS_VALUE

Undefined timestamp value.

Definition: avutil.h:248

filter_frame

static int filter_frame(AVFilterLink *inlink, AVFrame *frame)

Definition: af_adelay.c:326

AVFrame::format

int format

format of the frame, -1 if unknown or unset Values correspond to enum AVPixelFormat for video frames,...

Definition: frame.h:431

AudioDelayContext::input

AVFrame * input

Definition: af_adelay.c:51

FF_FILTER_FORWARD_WANTED

FF_FILTER_FORWARD_WANTED(outlink, inlink)

AV_SAMPLE_FMT_S16P

@ AV_SAMPLE_FMT_S16P

signed 16 bits, planar

Definition: samplefmt.h:64

internal.h

ChanDelay::index

size_t index

Definition: af_adelay.c:33

AVFrame::nb_samples

int nb_samples

number of audio samples (per channel) described by this frame

Definition: frame.h:424

#define i(width, name, range_min, range_max)

Definition: cbs_h2645.c:255

av_get_bytes_per_sample

int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt)

Return number of bytes per sample.

Definition: samplefmt.c:108

AudioDelayContext::chandelay

ChanDelay * chandelay

Definition: af_adelay.c:42

AVFrame::extended_data

uint8_t ** extended_data

pointers to the data planes/channels.

Definition: frame.h:405

av_malloc_array

#define av_malloc_array(a, b)

Definition: tableprint_vlc.h:31

ChanDelay::samples

uint8_t * samples

Definition: af_adelay.c:35

FFMIN

#define FFMIN(a, b)

Definition: macros.h:49

AVFilterPad::name

const char * name

Pad name.

Definition: internal.h:39

av_calloc

void * av_calloc(size_t nmemb, size_t size)

Definition: mem.c:262

CHANGE_DELAY

#define CHANGE_DELAY(name, type, fill)

Definition: af_adelay.c:104

AVFilter

Filter definition.

Definition: avfilter.h:166

ret

Definition: filter_design.txt:187

status

ov_status_e status

Definition: dnn_backend_openvino.c:120

adelay_inputs

static const AVFilterPad adelay_inputs[]

Definition: af_adelay.c:461

AV_OPT_FLAG_RUNTIME_PARAM

#define AV_OPT_FLAG_RUNTIME_PARAM

A generic parameter which can be set by the user at runtime.

Definition: opt.h:294

avfilter.h

process_command

static int process_command(AVFilterContext *ctx, const char *cmd, const char *args, char *res, int res_len, int flags)

Definition: af_adelay.c:272

AV_SAMPLE_FMT_DBLP

@ AV_SAMPLE_FMT_DBLP

double, planar

Definition: samplefmt.h:67

AudioDelayContext::eof

int eof

Definition: af_adelay.c:49

AVFilterLink::time_base

AVRational time_base

Define the time base used by the PTS of the frames/samples which will pass through this link.

Definition: avfilter.h:578

AVFilterContext

An instance of a filter.

Definition: avfilter.h:407

av_strdup

char * av_strdup(const char *s)

Duplicate a string.

Definition: mem.c:270

audio.h

AVFilterLink::ch_layout

AVChannelLayout ch_layout

channel layout of current buffer (see libavutil/channel_layout.h)

Definition: avfilter.h:569

AV_OPT_TYPE_BOOL

@ AV_OPT_TYPE_BOOL

Definition: opt.h:251

FILTER_OUTPUTS

#define FILTER_OUTPUTS(array)

Definition: internal.h:183

av_freep

#define av_freep(p)

Definition: tableprint_vlc.h:34

AudioDelayContext::all

int all

Definition: af_adelay.c:40

src

INIT_CLIP pixel * src

Definition: h264pred_template.c:418

Definition: ffmpeg_filter.c:409

uninit

static av_cold void uninit(AVFilterContext *ctx)

Definition: af_adelay.c:450

int32_t

Definition: audioconvert.c:56

AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL

#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:155

flags

#define flags(name, subs,...)

Definition: cbs_av1.c:482

av_log

#define av_log(a,...)

Definition: tableprint_vlc.h:27

ChanDelay::delay_index

size_t delay_index

Definition: af_adelay.c:32

ChanDelay::samples_size

unsigned int samples_size

Definition: af_adelay.c:34

avstring.h

AV_OPT_TYPE_STRING

@ AV_OPT_TYPE_STRING

Definition: opt.h:239

AudioDelayContext::max_delay

int64_t max_delay

Definition: af_adelay.c:46

int

Definition: ffmpeg_filter.c:409

AudioDelayContext::offset

int64_t offset

Definition: af_adelay.c:47

FILTER_SAMPLEFMTS

#define FILTER_SAMPLEFMTS(...)

Definition: internal.h:170

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