AVFrame: add side data.
[libav.git] / libavfilter / af_asyncts.c
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1/*
2 * This file is part of Libav.
3 *
4 * Libav 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 * Libav 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 Libav; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
19#include "libavresample/avresample.h"
20#include "libavutil/audio_fifo.h"
1d9c2dc8 21#include "libavutil/common.h"
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22#include "libavutil/mathematics.h"
23#include "libavutil/opt.h"
24#include "libavutil/samplefmt.h"
25
26#include "audio.h"
27#include "avfilter.h"
803391f7 28#include "internal.h"
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29
30typedef struct ASyncContext {
31 const AVClass *class;
32
33 AVAudioResampleContext *avr;
34 int64_t pts; ///< timestamp in samples of the first sample in fifo
35 int min_delta; ///< pad/trim min threshold in samples
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36 int first_frame; ///< 1 until filter_frame() has processed at least 1 frame with a pts != AV_NOPTS_VALUE
37 int64_t first_pts; ///< user-specified first expected pts, in samples
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38
39 /* options */
40 int resample;
41 float min_delta_sec;
42 int max_comp;
6f834293 43
565e4993 44 /* set by filter_frame() to signal an output frame to request_frame() */
6f834293 45 int got_output;
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46} ASyncContext;
47
48#define OFFSET(x) offsetof(ASyncContext, x)
49#define A AV_OPT_FLAG_AUDIO_PARAM
50static const AVOption options[] = {
e6153f17 51 { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, A },
9f26421b 52 { "min_delta", "Minimum difference between timestamps and audio data "
c7b610aa 53 "(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { .dbl = 0.1 }, 0, INT_MAX, A },
e6153f17 54 { "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { .i64 = 500 }, 0, INT_MAX, A },
c143de40 55 { "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 },
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56 { NULL },
57};
58
59static const AVClass async_class = {
60 .class_name = "asyncts filter",
61 .item_name = av_default_item_name,
62 .option = options,
63 .version = LIBAVUTIL_VERSION_INT,
64};
65
a5e8c41c 66static int init(AVFilterContext *ctx, const char *args)
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67{
68 ASyncContext *s = ctx->priv;
69 int ret;
70
71 s->class = &async_class;
72 av_opt_set_defaults(s);
73
74 if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {
75 av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
76 return ret;
77 }
78 av_opt_free(s);
79
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80 s->pts = AV_NOPTS_VALUE;
81 s->first_frame = 1;
82
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83 return 0;
84}
85
86static void uninit(AVFilterContext *ctx)
87{
88 ASyncContext *s = ctx->priv;
89
90 if (s->avr) {
91 avresample_close(s->avr);
92 avresample_free(&s->avr);
93 }
94}
95
96static int config_props(AVFilterLink *link)
97{
98 ASyncContext *s = link->src->priv;
99 int ret;
100
101 s->min_delta = s->min_delta_sec * link->sample_rate;
102 link->time_base = (AVRational){1, link->sample_rate};
103
104 s->avr = avresample_alloc_context();
105 if (!s->avr)
106 return AVERROR(ENOMEM);
107
108 av_opt_set_int(s->avr, "in_channel_layout", link->channel_layout, 0);
109 av_opt_set_int(s->avr, "out_channel_layout", link->channel_layout, 0);
110 av_opt_set_int(s->avr, "in_sample_fmt", link->format, 0);
111 av_opt_set_int(s->avr, "out_sample_fmt", link->format, 0);
112 av_opt_set_int(s->avr, "in_sample_rate", link->sample_rate, 0);
113 av_opt_set_int(s->avr, "out_sample_rate", link->sample_rate, 0);
114
115 if (s->resample)
116 av_opt_set_int(s->avr, "force_resampling", 1, 0);
117
118 if ((ret = avresample_open(s->avr)) < 0)
119 return ret;
120
121 return 0;
122}
123
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124/* get amount of data currently buffered, in samples */
125static int64_t get_delay(ASyncContext *s)
126{
127 return avresample_available(s->avr) + avresample_get_delay(s->avr);
128}
129
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130static void handle_trimming(AVFilterContext *ctx)
131{
132 ASyncContext *s = ctx->priv;
133
134 if (s->pts < s->first_pts) {
135 int delta = FFMIN(s->first_pts - s->pts, avresample_available(s->avr));
136 av_log(ctx, AV_LOG_VERBOSE, "Trimming %d samples from start\n",
137 delta);
138 avresample_read(s->avr, NULL, delta);
139 s->pts += delta;
140 } else if (s->first_frame)
141 s->pts = s->first_pts;
142}
143
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144static int request_frame(AVFilterLink *link)
145{
146 AVFilterContext *ctx = link->src;
147 ASyncContext *s = ctx->priv;
6f834293 148 int ret = 0;
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149 int nb_samples;
150
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151 s->got_output = 0;
152 while (ret >= 0 && !s->got_output)
153 ret = ff_request_frame(ctx->inputs[0]);
154
9f26421b 155 /* flush the fifo */
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156 if (ret == AVERROR_EOF) {
157 if (s->first_pts != AV_NOPTS_VALUE)
158 handle_trimming(ctx);
159
160 if (nb_samples = get_delay(s)) {
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161 AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
162 nb_samples);
163 if (!buf)
164 return AVERROR(ENOMEM);
165 ret = avresample_convert(s->avr, buf->extended_data,
166 buf->linesize[0], nb_samples, NULL, 0, 0);
167 if (ret <= 0) {
168 avfilter_unref_bufferp(&buf);
169 return (ret < 0) ? ret : AVERROR_EOF;
170 }
171
172 buf->pts = s->pts;
173 return ff_filter_frame(link, buf);
c143de40 174 }
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175 }
176
177 return ret;
178}
179
cd991462 180static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
9f26421b 181{
e7ba5b1d 182 int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
cd991462 183 buf->linesize[0], buf->audio->nb_samples);
9f26421b 184 avfilter_unref_buffer(buf);
cd991462 185 return ret;
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186}
187
565e4993 188static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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189{
190 AVFilterContext *ctx = inlink->dst;
191 ASyncContext *s = ctx->priv;
192 AVFilterLink *outlink = ctx->outputs[0];
193 int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
194 int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
195 av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
cd991462 196 int out_size, ret;
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197 int64_t delta;
198
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199 /* buffer data until we get the next timestamp */
200 if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) {
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201 if (pts != AV_NOPTS_VALUE) {
202 s->pts = pts - get_delay(s);
203 }
cd991462 204 return write_to_fifo(s, buf);
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205 }
206
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207 if (s->first_pts != AV_NOPTS_VALUE) {
208 handle_trimming(ctx);
209 if (!avresample_available(s->avr))
210 return write_to_fifo(s, buf);
211 }
212
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213 /* when we have two timestamps, compute how many samples would we have
214 * to add/remove to get proper sync between data and timestamps */
215 delta = pts - s->pts - get_delay(s);
216 out_size = avresample_available(s->avr);
217
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218 if (labs(delta) > s->min_delta ||
219 (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) {
9f26421b 220 av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
be51e589 221 out_size = av_clipl_int32((int64_t)out_size + delta);
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222 } else {
223 if (s->resample) {
224 int comp = av_clip(delta, -s->max_comp, s->max_comp);
225 av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
8083332c 226 avresample_set_compensation(s->avr, comp, inlink->sample_rate);
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227 }
228 delta = 0;
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229 }
230
231 if (out_size > 0) {
232 AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
233 out_size);
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234 if (!buf_out) {
235 ret = AVERROR(ENOMEM);
236 goto fail;
237 }
9f26421b 238
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239 if (s->first_frame && delta > 0) {
240 int ch;
241
242 av_samples_set_silence(buf_out->extended_data, 0, delta,
243 nb_channels, buf->format);
244
245 for (ch = 0; ch < nb_channels; ch++)
246 buf_out->extended_data[ch] += delta;
9f26421b 247
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248 avresample_read(s->avr, buf_out->extended_data, out_size);
249
250 for (ch = 0; ch < nb_channels; ch++)
251 buf_out->extended_data[ch] -= delta;
252 } else {
253 avresample_read(s->avr, buf_out->extended_data, out_size);
254
255 if (delta > 0) {
256 av_samples_set_silence(buf_out->extended_data, out_size - delta,
257 delta, nb_channels, buf->format);
258 }
9f26421b 259 }
c143de40 260 buf_out->pts = s->pts;
565e4993 261 ret = ff_filter_frame(outlink, buf_out);
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262 if (ret < 0)
263 goto fail;
6f834293 264 s->got_output = 1;
c143de40 265 } else if (avresample_available(s->avr)) {
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266 av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
267 "whole buffer.\n");
268 }
269
270 /* drain any remaining buffered data */
271 avresample_read(s->avr, NULL, avresample_available(s->avr));
272
273 s->pts = pts - avresample_get_delay(s->avr);
e7ba5b1d 274 ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
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275 buf->linesize[0], buf->audio->nb_samples);
276
c143de40 277 s->first_frame = 0;
cd991462 278fail:
9f26421b 279 avfilter_unref_buffer(buf);
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280
281 return ret;
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282}
283
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284static const AVFilterPad avfilter_af_asyncts_inputs[] = {
285 {
286 .name = "default",
287 .type = AVMEDIA_TYPE_AUDIO,
565e4993 288 .filter_frame = filter_frame,
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289 },
290 { NULL }
291};
292
293static const AVFilterPad avfilter_af_asyncts_outputs[] = {
294 {
295 .name = "default",
296 .type = AVMEDIA_TYPE_AUDIO,
297 .config_props = config_props,
298 .request_frame = request_frame
299 },
300 { NULL }
301};
302
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303AVFilter avfilter_af_asyncts = {
304 .name = "asyncts",
305 .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),
306
307 .init = init,
308 .uninit = uninit,
309
310 .priv_size = sizeof(ASyncContext),
311
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312 .inputs = avfilter_af_asyncts_inputs,
313 .outputs = avfilter_af_asyncts_outputs,
9f26421b 314};