1/* 2 * This file is part of FFmpeg. 3 * 4 * FFmpeg is free software; you can redistribute it and/or 5 * modify it under the terms of the GNU Lesser General Public 6 * License as published by the Free Software Foundation; either 7 * version 2.1 of the License, or (at your option) any later version. 8 * 9 * FFmpeg is distributed in the hope that it will be useful, 10 * but WITHOUT ANY WARRANTY; without even the implied warranty of 11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 12 * Lesser General Public License for more details. 13 * 14 * You should have received a copy of the GNU Lesser General Public 15 * License along with FFmpeg; if not, write to the Free Software 16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 17 */ 18 19#include <stdint.h> 20 21#include "libavresample/avresample.h" 22#include "libavutil/attributes.h" 23#include "libavutil/audio_fifo.h" 24#include "libavutil/common.h" 25#include "libavutil/mathematics.h" 26#include "libavutil/opt.h" 27#include "libavutil/samplefmt.h" 28 29#include "audio.h" 30#include "avfilter.h" 31#include "internal.h" 32 33typedef struct ASyncContext { 34 const AVClass *class; 35 36 AVAudioResampleContext *avr; 37 int64_t pts; ///< timestamp in samples of the first sample in fifo 38 int min_delta; ///< pad/trim min threshold in samples 39 int first_frame; ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE 40 int64_t first_pts; ///< user-specified first expected pts, in samples 41 int comp; ///< current resample compensation 42 43 /* options */ 44 int resample; 45 float min_delta_sec; 46 int max_comp; 47 48 /* set by filter_frame() to signal an output frame to request_frame() */ 49 int got_output; 50} ASyncContext; 51 52#define OFFSET(x) offsetof(ASyncContext, x) 53#define A AV_OPT_FLAG_AUDIO_PARAM 54#define F AV_OPT_FLAG_FILTERING_PARAM 55static const AVOption asyncts_options[] = { 56 { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, A|F }, 57 { "min_delta", "Minimum difference between timestamps and audio data " 58 "(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A|F }, 59 { "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A|F }, 60 { "first_pts", "Assume the first pts should be this value.", OFFSET(first_pts), AV_OPT_TYPE_INT64, { .i64 = AV_NOPTS_VALUE }, INT64_MIN, INT64_MAX, A|F }, 61 { NULL } 62}; 63 64AVFILTER_DEFINE_CLASS(asyncts); 65 66static av_cold int init(AVFilterContext *ctx) 67{ 68 ASyncContext *s = ctx->priv; 69 70 s->pts = AV_NOPTS_VALUE; 71 s->first_frame = 1; 72 73 return 0; 74} 75 76static av_cold void uninit(AVFilterContext *ctx) 77{ 78 ASyncContext *s = ctx->priv; 79 80 if (s->avr) { 81 avresample_close(s->avr); 82 avresample_free(&s->avr); 83 } 84} 85 86static int config_props(AVFilterLink *link) 87{ 88 ASyncContext *s = link->src->priv; 89 int ret; 90 91 s->min_delta = s->min_delta_sec * link->sample_rate; 92 link->time_base = (AVRational){1, link->sample_rate}; 93 94 s->avr = avresample_alloc_context(); 95 if (!s->avr) 96 return AVERROR(ENOMEM); 97 98 av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0); 99 av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0); 100 av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0); 101 av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0); 102 av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0); 103 av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0); 104 105 if (s->resample) 106 av_opt_set_int(s->avr, "force_resampling", 1, 0); 107 108 if ((ret = avresample_open(s->avr)) < 0) 109 return ret; 110 111 return 0; 112} 113 114/* get amount of data currently buffered, in samples */ 115static int64_t get_delay(ASyncContext *s) 116{ 117 return avresample_available(s->avr) + avresample_get_delay(s->avr); 118} 119 120static void handle_trimming(AVFilterContext *ctx) 121{ 122 ASyncContext *s = ctx->priv; 123 124 if (s->pts < s->first_pts) { 125 int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr)); 126 av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n", 127 delta); 128 avresample_read(s->avr, NULL, delta); 129 s->pts += delta; 130 } else if (s->first_frame) 131 s->pts = s->first_pts; 132} 133 134static int request_frame(AVFilterLink *link) 135{ 136 AVFilterContext *ctx = link->src; 137 ASyncContext *s = ctx->priv; 138 int ret = 0; 139 int nb_samples; 140 141 s->got_output = 0; 142 while (ret >= 0 && !s->got_output) 143 ret = ff_request_frame(ctx->inputs[0]); 144 145 /* flush the fifo */ 146 if (ret == AVERROR_EOF) { 147 if (s->first_pts != AV_NOPTS_VALUE) 148 handle_trimming(ctx); 149 150 if (nb_samples = get_delay(s)) { 151 AVFrame *buf = ff_get_audio_buffer(link, nb_samples); 152 if (!buf) 153 return AVERROR(ENOMEM); 154 ret = avresample_convert(s->avr, buf->extended_data, 155 buf->linesize[0], nb_samples, NULL, 0, 0); 156 if (ret <= 0) { 157 av_frame_free(&buf); 158 return (ret < 0) ? ret : AVERROR_EOF; 159 } 160 161 buf->pts = s->pts; 162 return ff_filter_frame(link, buf); 163 } 164 } 165 166 return ret; 167} 168 169static int write_to_fifo(ASyncContext *s, AVFrame *buf) 170{ 171 int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, 172 buf->linesize[0], buf->nb_samples); 173 av_frame_free(&buf); 174 return ret; 175} 176 177static int filter_frame(AVFilterLink *inlink, AVFrame *buf) 178{ 179 AVFilterContext *ctx = inlink->dst; 180 ASyncContext *s = ctx->priv; 181 AVFilterLink *outlink = ctx->outputs[0]; 182 int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout); 183 int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts : 184 av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); 185 int out_size, ret; 186 int64_t delta; 187 int64_t new_pts; 188 189 /* buffer data until we get the next timestamp */ 190 if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) { 191 if (pts != AV_NOPTS_VALUE) { 192 s->pts = pts - get_delay(s); 193 } 194 return write_to_fifo(s, buf); 195 } 196 197 if (s->first_pts != AV_NOPTS_VALUE) { 198 handle_trimming(ctx); 199 if (!avresample_available(s->avr)) 200 return write_to_fifo(s, buf); 201 } 202 203 /* when we have two timestamps, compute how many samples would we have 204 * to add/remove to get proper sync between data and timestamps */ 205 delta = pts - s->pts - get_delay(s); 206 out_size = avresample_available(s->avr); 207 208 if (labs(delta) > s->min_delta || 209 (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) { 210 av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta); 211 out_size = av_clipl_int32((int64_t)out_size + delta); 212 } else { 213 if (s->resample) { 214 // adjust the compensation if delta is non-zero 215 int delay = get_delay(s); 216 int comp = s->comp + av_clip(delta * inlink->sample_rate / delay, 217 -s->max_comp, s->max_comp); 218 if (comp != s->comp) { 219 av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp); 220 if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) { 221 s->comp = comp; 222 } 223 } 224 } 225 // adjust PTS to avoid monotonicity errors with input PTS jitter 226 pts -= delta; 227 delta = 0; 228 } 229 230 if (out_size > 0) { 231 AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size); 232 if (!buf_out) { 233 ret = AVERROR(ENOMEM); 234 goto fail; 235 } 236 237 if (s->first_frame && delta > 0) { 238 int planar = av_sample_fmt_is_planar(buf_out->format); 239 int planes = planar ? nb_channels : 1; 240 int block_size = av_get_bytes_per_sample(buf_out->format) * 241 (planar ? 1 : nb_channels); 242 243 int ch; 244 245 av_samples_set_silence(buf_out->extended_data, 0, delta, 246 nb_channels, buf->format); 247 248 for (ch = 0; ch < planes; ch++) 249 buf_out->extended_data[ch] += delta * block_size; 250 251 avresample_read(s->avr, buf_out->extended_data, out_size); 252 253 for (ch = 0; ch < planes; ch++) 254 buf_out->extended_data[ch] -= delta * block_size; 255 } else { 256 avresample_read(s->avr, buf_out->extended_data, out_size); 257 258 if (delta > 0) { 259 av_samples_set_silence(buf_out->extended_data, out_size - delta, 260 delta, nb_channels, buf->format); 261 } 262 } 263 buf_out->pts = s->pts; 264 ret = ff_filter_frame(outlink, buf_out); 265 if (ret < 0) 266 goto fail; 267 s->got_output = 1; 268 } else if (avresample_available(s->avr)) { 269 av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " 270 "whole buffer.\n"); 271 } 272 273 /* drain any remaining buffered data */ 274 avresample_read(s->avr, NULL, avresample_available(s->avr)); 275 276 new_pts = pts - avresample_get_delay(s->avr); 277 /* check for s->pts monotonicity */ 278 if (new_pts > s->pts) { 279 s->pts = new_pts; 280 ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, 281 buf->linesize[0], buf->nb_samples); 282 } else { 283 av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " 284 "whole buffer.\n"); 285 ret = 0; 286 } 287 288 s->first_frame = 0; 289fail: 290 av_frame_free(&buf); 291 292 return ret; 293} 294 295static const AVFilterPad avfilter_af_asyncts_inputs[] = { 296 { 297 .name = "default", 298 .type = AVMEDIA_TYPE_AUDIO, 299 .filter_frame = filter_frame 300 }, 301 { NULL } 302}; 303 304static const AVFilterPad avfilter_af_asyncts_outputs[] = { 305 { 306 .name = "default", 307 .type = AVMEDIA_TYPE_AUDIO, 308 .config_props = config_props, 309 .request_frame = request_frame 310 }, 311 { NULL } 312}; 313 314AVFilter ff_af_asyncts = { 315 .name = "asyncts", 316 .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"), 317 .init = init, 318 .uninit = uninit, 319 .priv_size = sizeof(ASyncContext), 320 .priv_class = &asyncts_class, 321 .inputs = avfilter_af_asyncts_inputs, 322 .outputs = avfilter_af_asyncts_outputs, 323}; 324