| 1 | /* |
| 2 | * Audio Mix Filter |
| 3 | * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> |
| 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 | /** |
| 23 | * @file |
| 24 | * Audio Mix Filter |
| 25 | * |
| 26 | * Mixes audio from multiple sources into a single output. The channel layout, |
| 27 | * sample rate, and sample format will be the same for all inputs and the |
| 28 | * output. |
| 29 | */ |
| 30 | |
| 31 | #include "libavutil/attributes.h" |
| 32 | #include "libavutil/audio_fifo.h" |
| 33 | #include "libavutil/avassert.h" |
| 34 | #include "libavutil/avstring.h" |
| 35 | #include "libavutil/channel_layout.h" |
| 36 | #include "libavutil/common.h" |
| 37 | #include "libavutil/float_dsp.h" |
| 38 | #include "libavutil/mathematics.h" |
| 39 | #include "libavutil/opt.h" |
| 40 | #include "libavutil/samplefmt.h" |
| 41 | |
| 42 | #include "audio.h" |
| 43 | #include "avfilter.h" |
| 44 | #include "formats.h" |
| 45 | #include "internal.h" |
| 46 | |
| 47 | #define INPUT_OFF 0 /**< input has reached EOF */ |
| 48 | #define INPUT_ON 1 /**< input is active */ |
| 49 | #define INPUT_INACTIVE 2 /**< input is on, but is currently inactive */ |
| 50 | |
| 51 | #define DURATION_LONGEST 0 |
| 52 | #define DURATION_SHORTEST 1 |
| 53 | #define DURATION_FIRST 2 |
| 54 | |
| 55 | |
| 56 | typedef struct FrameInfo { |
| 57 | int nb_samples; |
| 58 | int64_t pts; |
| 59 | struct FrameInfo *next; |
| 60 | } FrameInfo; |
| 61 | |
| 62 | /** |
| 63 | * Linked list used to store timestamps and frame sizes of all frames in the |
| 64 | * FIFO for the first input. |
| 65 | * |
| 66 | * This is needed to keep timestamps synchronized for the case where multiple |
| 67 | * input frames are pushed to the filter for processing before a frame is |
| 68 | * requested by the output link. |
| 69 | */ |
| 70 | typedef struct FrameList { |
| 71 | int nb_frames; |
| 72 | int nb_samples; |
| 73 | FrameInfo *list; |
| 74 | FrameInfo *end; |
| 75 | } FrameList; |
| 76 | |
| 77 | static void frame_list_clear(FrameList *frame_list) |
| 78 | { |
| 79 | if (frame_list) { |
| 80 | while (frame_list->list) { |
| 81 | FrameInfo *info = frame_list->list; |
| 82 | frame_list->list = info->next; |
| 83 | av_free(info); |
| 84 | } |
| 85 | frame_list->nb_frames = 0; |
| 86 | frame_list->nb_samples = 0; |
| 87 | frame_list->end = NULL; |
| 88 | } |
| 89 | } |
| 90 | |
| 91 | static int frame_list_next_frame_size(FrameList *frame_list) |
| 92 | { |
| 93 | if (!frame_list->list) |
| 94 | return 0; |
| 95 | return frame_list->list->nb_samples; |
| 96 | } |
| 97 | |
| 98 | static int64_t frame_list_next_pts(FrameList *frame_list) |
| 99 | { |
| 100 | if (!frame_list->list) |
| 101 | return AV_NOPTS_VALUE; |
| 102 | return frame_list->list->pts; |
| 103 | } |
| 104 | |
| 105 | static void frame_list_remove_samples(FrameList *frame_list, int nb_samples) |
| 106 | { |
| 107 | if (nb_samples >= frame_list->nb_samples) { |
| 108 | frame_list_clear(frame_list); |
| 109 | } else { |
| 110 | int samples = nb_samples; |
| 111 | while (samples > 0) { |
| 112 | FrameInfo *info = frame_list->list; |
| 113 | av_assert0(info); |
| 114 | if (info->nb_samples <= samples) { |
| 115 | samples -= info->nb_samples; |
| 116 | frame_list->list = info->next; |
| 117 | if (!frame_list->list) |
| 118 | frame_list->end = NULL; |
| 119 | frame_list->nb_frames--; |
| 120 | frame_list->nb_samples -= info->nb_samples; |
| 121 | av_free(info); |
| 122 | } else { |
| 123 | info->nb_samples -= samples; |
| 124 | info->pts += samples; |
| 125 | frame_list->nb_samples -= samples; |
| 126 | samples = 0; |
| 127 | } |
| 128 | } |
| 129 | } |
| 130 | } |
| 131 | |
| 132 | static int frame_list_add_frame(FrameList *frame_list, int nb_samples, int64_t pts) |
| 133 | { |
| 134 | FrameInfo *info = av_malloc(sizeof(*info)); |
| 135 | if (!info) |
| 136 | return AVERROR(ENOMEM); |
| 137 | info->nb_samples = nb_samples; |
| 138 | info->pts = pts; |
| 139 | info->next = NULL; |
| 140 | |
| 141 | if (!frame_list->list) { |
| 142 | frame_list->list = info; |
| 143 | frame_list->end = info; |
| 144 | } else { |
| 145 | av_assert0(frame_list->end); |
| 146 | frame_list->end->next = info; |
| 147 | frame_list->end = info; |
| 148 | } |
| 149 | frame_list->nb_frames++; |
| 150 | frame_list->nb_samples += nb_samples; |
| 151 | |
| 152 | return 0; |
| 153 | } |
| 154 | |
| 155 | |
| 156 | typedef struct MixContext { |
| 157 | const AVClass *class; /**< class for AVOptions */ |
| 158 | AVFloatDSPContext *fdsp; |
| 159 | |
| 160 | int nb_inputs; /**< number of inputs */ |
| 161 | int active_inputs; /**< number of input currently active */ |
| 162 | int duration_mode; /**< mode for determining duration */ |
| 163 | float dropout_transition; /**< transition time when an input drops out */ |
| 164 | |
| 165 | int nb_channels; /**< number of channels */ |
| 166 | int sample_rate; /**< sample rate */ |
| 167 | int planar; |
| 168 | AVAudioFifo **fifos; /**< audio fifo for each input */ |
| 169 | uint8_t *input_state; /**< current state of each input */ |
| 170 | float *input_scale; /**< mixing scale factor for each input */ |
| 171 | float scale_norm; /**< normalization factor for all inputs */ |
| 172 | int64_t next_pts; /**< calculated pts for next output frame */ |
| 173 | FrameList *frame_list; /**< list of frame info for the first input */ |
| 174 | } MixContext; |
| 175 | |
| 176 | #define OFFSET(x) offsetof(MixContext, x) |
| 177 | #define A AV_OPT_FLAG_AUDIO_PARAM |
| 178 | #define F AV_OPT_FLAG_FILTERING_PARAM |
| 179 | static const AVOption amix_options[] = { |
| 180 | { "inputs", "Number of inputs.", |
| 181 | OFFSET(nb_inputs), AV_OPT_TYPE_INT, { .i64 = 2 }, 1, 32, A|F }, |
| 182 | { "duration", "How to determine the end-of-stream.", |
| 183 | OFFSET(duration_mode), AV_OPT_TYPE_INT, { .i64 = DURATION_LONGEST }, 0, 2, A|F, "duration" }, |
| 184 | { "longest", "Duration of longest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_LONGEST }, INT_MIN, INT_MAX, A|F, "duration" }, |
| 185 | { "shortest", "Duration of shortest input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_SHORTEST }, INT_MIN, INT_MAX, A|F, "duration" }, |
| 186 | { "first", "Duration of first input.", 0, AV_OPT_TYPE_CONST, { .i64 = DURATION_FIRST }, INT_MIN, INT_MAX, A|F, "duration" }, |
| 187 | { "dropout_transition", "Transition time, in seconds, for volume " |
| 188 | "renormalization when an input stream ends.", |
| 189 | OFFSET(dropout_transition), AV_OPT_TYPE_FLOAT, { .dbl = 2.0 }, 0, INT_MAX, A|F }, |
| 190 | { NULL } |
| 191 | }; |
| 192 | |
| 193 | AVFILTER_DEFINE_CLASS(amix); |
| 194 | |
| 195 | /** |
| 196 | * Update the scaling factors to apply to each input during mixing. |
| 197 | * |
| 198 | * This balances the full volume range between active inputs and handles |
| 199 | * volume transitions when EOF is encountered on an input but mixing continues |
| 200 | * with the remaining inputs. |
| 201 | */ |
| 202 | static void calculate_scales(MixContext *s, int nb_samples) |
| 203 | { |
| 204 | int i; |
| 205 | |
| 206 | if (s->scale_norm > s->active_inputs) { |
| 207 | s->scale_norm -= nb_samples / (s->dropout_transition * s->sample_rate); |
| 208 | s->scale_norm = FFMAX(s->scale_norm, s->active_inputs); |
| 209 | } |
| 210 | |
| 211 | for (i = 0; i < s->nb_inputs; i++) { |
| 212 | if (s->input_state[i] == INPUT_ON) |
| 213 | s->input_scale[i] = 1.0f / s->scale_norm; |
| 214 | else |
| 215 | s->input_scale[i] = 0.0f; |
| 216 | } |
| 217 | } |
| 218 | |
| 219 | static int config_output(AVFilterLink *outlink) |
| 220 | { |
| 221 | AVFilterContext *ctx = outlink->src; |
| 222 | MixContext *s = ctx->priv; |
| 223 | int i; |
| 224 | char buf[64]; |
| 225 | |
| 226 | s->planar = av_sample_fmt_is_planar(outlink->format); |
| 227 | s->sample_rate = outlink->sample_rate; |
| 228 | outlink->time_base = (AVRational){ 1, outlink->sample_rate }; |
| 229 | s->next_pts = AV_NOPTS_VALUE; |
| 230 | |
| 231 | s->frame_list = av_mallocz(sizeof(*s->frame_list)); |
| 232 | if (!s->frame_list) |
| 233 | return AVERROR(ENOMEM); |
| 234 | |
| 235 | s->fifos = av_mallocz(s->nb_inputs * sizeof(*s->fifos)); |
| 236 | if (!s->fifos) |
| 237 | return AVERROR(ENOMEM); |
| 238 | |
| 239 | s->nb_channels = av_get_channel_layout_nb_channels(outlink->channel_layout); |
| 240 | for (i = 0; i < s->nb_inputs; i++) { |
| 241 | s->fifos[i] = av_audio_fifo_alloc(outlink->format, s->nb_channels, 1024); |
| 242 | if (!s->fifos[i]) |
| 243 | return AVERROR(ENOMEM); |
| 244 | } |
| 245 | |
| 246 | s->input_state = av_malloc(s->nb_inputs); |
| 247 | if (!s->input_state) |
| 248 | return AVERROR(ENOMEM); |
| 249 | memset(s->input_state, INPUT_ON, s->nb_inputs); |
| 250 | s->active_inputs = s->nb_inputs; |
| 251 | |
| 252 | s->input_scale = av_mallocz_array(s->nb_inputs, sizeof(*s->input_scale)); |
| 253 | if (!s->input_scale) |
| 254 | return AVERROR(ENOMEM); |
| 255 | s->scale_norm = s->active_inputs; |
| 256 | calculate_scales(s, 0); |
| 257 | |
| 258 | av_get_channel_layout_string(buf, sizeof(buf), -1, outlink->channel_layout); |
| 259 | |
| 260 | av_log(ctx, AV_LOG_VERBOSE, |
| 261 | "inputs:%d fmt:%s srate:%d cl:%s\n", s->nb_inputs, |
| 262 | av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf); |
| 263 | |
| 264 | return 0; |
| 265 | } |
| 266 | |
| 267 | /** |
| 268 | * Read samples from the input FIFOs, mix, and write to the output link. |
| 269 | */ |
| 270 | static int output_frame(AVFilterLink *outlink, int nb_samples) |
| 271 | { |
| 272 | AVFilterContext *ctx = outlink->src; |
| 273 | MixContext *s = ctx->priv; |
| 274 | AVFrame *out_buf, *in_buf; |
| 275 | int i; |
| 276 | |
| 277 | calculate_scales(s, nb_samples); |
| 278 | |
| 279 | out_buf = ff_get_audio_buffer(outlink, nb_samples); |
| 280 | if (!out_buf) |
| 281 | return AVERROR(ENOMEM); |
| 282 | |
| 283 | in_buf = ff_get_audio_buffer(outlink, nb_samples); |
| 284 | if (!in_buf) { |
| 285 | av_frame_free(&out_buf); |
| 286 | return AVERROR(ENOMEM); |
| 287 | } |
| 288 | |
| 289 | for (i = 0; i < s->nb_inputs; i++) { |
| 290 | if (s->input_state[i] == INPUT_ON) { |
| 291 | int planes, plane_size, p; |
| 292 | |
| 293 | av_audio_fifo_read(s->fifos[i], (void **)in_buf->extended_data, |
| 294 | nb_samples); |
| 295 | |
| 296 | planes = s->planar ? s->nb_channels : 1; |
| 297 | plane_size = nb_samples * (s->planar ? 1 : s->nb_channels); |
| 298 | plane_size = FFALIGN(plane_size, 16); |
| 299 | |
| 300 | for (p = 0; p < planes; p++) { |
| 301 | s->fdsp->vector_fmac_scalar((float *)out_buf->extended_data[p], |
| 302 | (float *) in_buf->extended_data[p], |
| 303 | s->input_scale[i], plane_size); |
| 304 | } |
| 305 | } |
| 306 | } |
| 307 | av_frame_free(&in_buf); |
| 308 | |
| 309 | out_buf->pts = s->next_pts; |
| 310 | if (s->next_pts != AV_NOPTS_VALUE) |
| 311 | s->next_pts += nb_samples; |
| 312 | |
| 313 | return ff_filter_frame(outlink, out_buf); |
| 314 | } |
| 315 | |
| 316 | /** |
| 317 | * Returns the smallest number of samples available in the input FIFOs other |
| 318 | * than that of the first input. |
| 319 | */ |
| 320 | static int get_available_samples(MixContext *s) |
| 321 | { |
| 322 | int i; |
| 323 | int available_samples = INT_MAX; |
| 324 | |
| 325 | av_assert0(s->nb_inputs > 1); |
| 326 | |
| 327 | for (i = 1; i < s->nb_inputs; i++) { |
| 328 | int nb_samples; |
| 329 | if (s->input_state[i] == INPUT_OFF) |
| 330 | continue; |
| 331 | nb_samples = av_audio_fifo_size(s->fifos[i]); |
| 332 | available_samples = FFMIN(available_samples, nb_samples); |
| 333 | } |
| 334 | if (available_samples == INT_MAX) |
| 335 | return 0; |
| 336 | return available_samples; |
| 337 | } |
| 338 | |
| 339 | /** |
| 340 | * Requests a frame, if needed, from each input link other than the first. |
| 341 | */ |
| 342 | static int request_samples(AVFilterContext *ctx, int min_samples) |
| 343 | { |
| 344 | MixContext *s = ctx->priv; |
| 345 | int i, ret; |
| 346 | |
| 347 | av_assert0(s->nb_inputs > 1); |
| 348 | |
| 349 | for (i = 1; i < s->nb_inputs; i++) { |
| 350 | ret = 0; |
| 351 | if (s->input_state[i] == INPUT_OFF) |
| 352 | continue; |
| 353 | while (!ret && av_audio_fifo_size(s->fifos[i]) < min_samples) |
| 354 | ret = ff_request_frame(ctx->inputs[i]); |
| 355 | if (ret == AVERROR_EOF) { |
| 356 | if (av_audio_fifo_size(s->fifos[i]) == 0) { |
| 357 | s->input_state[i] = INPUT_OFF; |
| 358 | continue; |
| 359 | } |
| 360 | } else if (ret < 0) |
| 361 | return ret; |
| 362 | } |
| 363 | return 0; |
| 364 | } |
| 365 | |
| 366 | /** |
| 367 | * Calculates the number of active inputs and determines EOF based on the |
| 368 | * duration option. |
| 369 | * |
| 370 | * @return 0 if mixing should continue, or AVERROR_EOF if mixing should stop. |
| 371 | */ |
| 372 | static int calc_active_inputs(MixContext *s) |
| 373 | { |
| 374 | int i; |
| 375 | int active_inputs = 0; |
| 376 | for (i = 0; i < s->nb_inputs; i++) |
| 377 | active_inputs += !!(s->input_state[i] != INPUT_OFF); |
| 378 | s->active_inputs = active_inputs; |
| 379 | |
| 380 | if (!active_inputs || |
| 381 | (s->duration_mode == DURATION_FIRST && s->input_state[0] == INPUT_OFF) || |
| 382 | (s->duration_mode == DURATION_SHORTEST && active_inputs != s->nb_inputs)) |
| 383 | return AVERROR_EOF; |
| 384 | return 0; |
| 385 | } |
| 386 | |
| 387 | static int request_frame(AVFilterLink *outlink) |
| 388 | { |
| 389 | AVFilterContext *ctx = outlink->src; |
| 390 | MixContext *s = ctx->priv; |
| 391 | int ret; |
| 392 | int wanted_samples, available_samples; |
| 393 | |
| 394 | ret = calc_active_inputs(s); |
| 395 | if (ret < 0) |
| 396 | return ret; |
| 397 | |
| 398 | if (s->input_state[0] == INPUT_OFF) { |
| 399 | ret = request_samples(ctx, 1); |
| 400 | if (ret < 0) |
| 401 | return ret; |
| 402 | |
| 403 | ret = calc_active_inputs(s); |
| 404 | if (ret < 0) |
| 405 | return ret; |
| 406 | |
| 407 | available_samples = get_available_samples(s); |
| 408 | if (!available_samples) |
| 409 | return AVERROR(EAGAIN); |
| 410 | |
| 411 | return output_frame(outlink, available_samples); |
| 412 | } |
| 413 | |
| 414 | if (s->frame_list->nb_frames == 0) { |
| 415 | ret = ff_request_frame(ctx->inputs[0]); |
| 416 | if (ret == AVERROR_EOF) { |
| 417 | s->input_state[0] = INPUT_OFF; |
| 418 | if (s->nb_inputs == 1) |
| 419 | return AVERROR_EOF; |
| 420 | else |
| 421 | return AVERROR(EAGAIN); |
| 422 | } else if (ret < 0) |
| 423 | return ret; |
| 424 | } |
| 425 | av_assert0(s->frame_list->nb_frames > 0); |
| 426 | |
| 427 | wanted_samples = frame_list_next_frame_size(s->frame_list); |
| 428 | |
| 429 | if (s->active_inputs > 1) { |
| 430 | ret = request_samples(ctx, wanted_samples); |
| 431 | if (ret < 0) |
| 432 | return ret; |
| 433 | |
| 434 | ret = calc_active_inputs(s); |
| 435 | if (ret < 0) |
| 436 | return ret; |
| 437 | } |
| 438 | |
| 439 | if (s->active_inputs > 1) { |
| 440 | available_samples = get_available_samples(s); |
| 441 | if (!available_samples) |
| 442 | return AVERROR(EAGAIN); |
| 443 | available_samples = FFMIN(available_samples, wanted_samples); |
| 444 | } else { |
| 445 | available_samples = wanted_samples; |
| 446 | } |
| 447 | |
| 448 | s->next_pts = frame_list_next_pts(s->frame_list); |
| 449 | frame_list_remove_samples(s->frame_list, available_samples); |
| 450 | |
| 451 | return output_frame(outlink, available_samples); |
| 452 | } |
| 453 | |
| 454 | static int filter_frame(AVFilterLink *inlink, AVFrame *buf) |
| 455 | { |
| 456 | AVFilterContext *ctx = inlink->dst; |
| 457 | MixContext *s = ctx->priv; |
| 458 | AVFilterLink *outlink = ctx->outputs[0]; |
| 459 | int i, ret = 0; |
| 460 | |
| 461 | for (i = 0; i < ctx->nb_inputs; i++) |
| 462 | if (ctx->inputs[i] == inlink) |
| 463 | break; |
| 464 | if (i >= ctx->nb_inputs) { |
| 465 | av_log(ctx, AV_LOG_ERROR, "unknown input link\n"); |
| 466 | ret = AVERROR(EINVAL); |
| 467 | goto fail; |
| 468 | } |
| 469 | |
| 470 | if (i == 0) { |
| 471 | int64_t pts = av_rescale_q(buf->pts, inlink->time_base, |
| 472 | outlink->time_base); |
| 473 | ret = frame_list_add_frame(s->frame_list, buf->nb_samples, pts); |
| 474 | if (ret < 0) |
| 475 | goto fail; |
| 476 | } |
| 477 | |
| 478 | ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data, |
| 479 | buf->nb_samples); |
| 480 | |
| 481 | fail: |
| 482 | av_frame_free(&buf); |
| 483 | |
| 484 | return ret; |
| 485 | } |
| 486 | |
| 487 | static av_cold int init(AVFilterContext *ctx) |
| 488 | { |
| 489 | MixContext *s = ctx->priv; |
| 490 | int i; |
| 491 | |
| 492 | for (i = 0; i < s->nb_inputs; i++) { |
| 493 | char name[32]; |
| 494 | AVFilterPad pad = { 0 }; |
| 495 | |
| 496 | snprintf(name, sizeof(name), "input%d", i); |
| 497 | pad.type = AVMEDIA_TYPE_AUDIO; |
| 498 | pad.name = av_strdup(name); |
| 499 | if (!pad.name) |
| 500 | return AVERROR(ENOMEM); |
| 501 | pad.filter_frame = filter_frame; |
| 502 | |
| 503 | ff_insert_inpad(ctx, i, &pad); |
| 504 | } |
| 505 | |
| 506 | s->fdsp = avpriv_float_dsp_alloc(0); |
| 507 | if (!s->fdsp) |
| 508 | return AVERROR(ENOMEM); |
| 509 | |
| 510 | return 0; |
| 511 | } |
| 512 | |
| 513 | static av_cold void uninit(AVFilterContext *ctx) |
| 514 | { |
| 515 | int i; |
| 516 | MixContext *s = ctx->priv; |
| 517 | |
| 518 | if (s->fifos) { |
| 519 | for (i = 0; i < s->nb_inputs; i++) |
| 520 | av_audio_fifo_free(s->fifos[i]); |
| 521 | av_freep(&s->fifos); |
| 522 | } |
| 523 | frame_list_clear(s->frame_list); |
| 524 | av_freep(&s->frame_list); |
| 525 | av_freep(&s->input_state); |
| 526 | av_freep(&s->input_scale); |
| 527 | av_freep(&s->fdsp); |
| 528 | |
| 529 | for (i = 0; i < ctx->nb_inputs; i++) |
| 530 | av_freep(&ctx->input_pads[i].name); |
| 531 | } |
| 532 | |
| 533 | static int query_formats(AVFilterContext *ctx) |
| 534 | { |
| 535 | AVFilterFormats *formats = NULL; |
| 536 | AVFilterChannelLayouts *layouts; |
| 537 | |
| 538 | layouts = ff_all_channel_layouts(); |
| 539 | |
| 540 | if (!layouts) |
| 541 | return AVERROR(ENOMEM); |
| 542 | |
| 543 | ff_add_format(&formats, AV_SAMPLE_FMT_FLT); |
| 544 | ff_add_format(&formats, AV_SAMPLE_FMT_FLTP); |
| 545 | ff_set_common_formats(ctx, formats); |
| 546 | ff_set_common_channel_layouts(ctx, layouts); |
| 547 | ff_set_common_samplerates(ctx, ff_all_samplerates()); |
| 548 | return 0; |
| 549 | } |
| 550 | |
| 551 | static const AVFilterPad avfilter_af_amix_outputs[] = { |
| 552 | { |
| 553 | .name = "default", |
| 554 | .type = AVMEDIA_TYPE_AUDIO, |
| 555 | .config_props = config_output, |
| 556 | .request_frame = request_frame |
| 557 | }, |
| 558 | { NULL } |
| 559 | }; |
| 560 | |
| 561 | AVFilter ff_af_amix = { |
| 562 | .name = "amix", |
| 563 | .description = NULL_IF_CONFIG_SMALL("Audio mixing."), |
| 564 | .priv_size = sizeof(MixContext), |
| 565 | .priv_class = &amix_class, |
| 566 | .init = init, |
| 567 | .uninit = uninit, |
| 568 | .query_formats = query_formats, |
| 569 | .inputs = NULL, |
| 570 | .outputs = avfilter_af_amix_outputs, |
| 571 | .flags = AVFILTER_FLAG_DYNAMIC_INPUTS, |
| 572 | }; |