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af_asyncts.c
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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 
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 
33 typedef struct ASyncContext {
34  const AVClass *class;
35 
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;
46  int max_comp;
47 
48  /* set by filter_frame() to signal an output frame to request_frame() */
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
55 static 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 
64 AVFILTER_DEFINE_CLASS(asyncts);
65 
66 static 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 
76 static av_cold void uninit(AVFilterContext *ctx)
77 {
78  ASyncContext *s = ctx->priv;
79 
80  if (s->avr) {
82  avresample_free(&s->avr);
83  }
84 }
85 
86 static 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 
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 */
115 static int64_t get_delay(ASyncContext *s)
116 {
118 }
119 
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 
134 static 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 
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 
177 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
178 {
179  AVFilterContext *ctx = inlink->dst;
180  ASyncContext *s = ctx->priv;
181  AVFilterLink *outlink = ctx->outputs[0];
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 */
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;
289 fail:
290  av_frame_free(&buf);
291 
292  return ret;
293 }
294 
296  {
297  .name = "default",
298  .type = AVMEDIA_TYPE_AUDIO,
299  .filter_frame = filter_frame
300  },
301  { NULL }
302 };
303 
305  {
306  .name = "default",
307  .type = AVMEDIA_TYPE_AUDIO,
308  .config_props = config_props,
309  .request_frame = request_frame
310  },
311  { NULL }
312 };
313 
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 };