2 * This file is part of FFmpeg.
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.
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.
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
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"
33 typedef struct ASyncContext {
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
48 /* set by filter_frame() to signal an output frame to request_frame() */
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_BOOL, { .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 },
64 AVFILTER_DEFINE_CLASS(asyncts);
66 static av_cold int init(AVFilterContext *ctx)
68 ASyncContext *s = ctx->priv;
70 s->pts = AV_NOPTS_VALUE;
76 static av_cold void uninit(AVFilterContext *ctx)
78 ASyncContext *s = ctx->priv;
81 avresample_close(s->avr);
82 avresample_free(&s->avr);
86 static int config_props(AVFilterLink *link)
88 ASyncContext *s = link->src->priv;
91 s->min_delta = s->min_delta_sec * link->sample_rate;
92 link->time_base = (AVRational){1, link->sample_rate};
94 s->avr = avresample_alloc_context();
96 return AVERROR(ENOMEM);
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);
106 av_opt_set_int(s->avr, "force_resampling", 1, 0);
108 if ((ret = avresample_open(s->avr)) < 0)
114 /* get amount of data currently buffered, in samples */
115 static int64_t get_delay(ASyncContext *s)
117 return avresample_available(s->avr) + avresample_get_delay(s->avr);
120 static void handle_trimming(AVFilterContext *ctx)
122 ASyncContext *s = ctx->priv;
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",
128 avresample_read(s->avr, NULL, delta);
130 } else if (s->first_frame)
131 s->pts = s->first_pts;
134 static int request_frame(AVFilterLink *link)
136 AVFilterContext *ctx = link->src;
137 ASyncContext *s = ctx->priv;
142 ret = ff_request_frame(ctx->inputs[0]);
145 if (ret == AVERROR_EOF) {
146 if (s->first_pts != AV_NOPTS_VALUE)
147 handle_trimming(ctx);
149 if (nb_samples = get_delay(s)) {
150 AVFrame *buf = ff_get_audio_buffer(link, nb_samples);
152 return AVERROR(ENOMEM);
153 ret = avresample_convert(s->avr, buf->extended_data,
154 buf->linesize[0], nb_samples, NULL, 0, 0);
157 return (ret < 0) ? ret : AVERROR_EOF;
161 return ff_filter_frame(link, buf);
168 static int write_to_fifo(ASyncContext *s, AVFrame *buf)
170 int ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
171 buf->linesize[0], buf->nb_samples);
176 static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
178 AVFilterContext *ctx = inlink->dst;
179 ASyncContext *s = ctx->priv;
180 AVFilterLink *outlink = ctx->outputs[0];
181 int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout);
182 int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
183 av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
188 /* buffer data until we get the next timestamp */
189 if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) {
190 if (pts != AV_NOPTS_VALUE) {
191 s->pts = pts - get_delay(s);
193 return write_to_fifo(s, buf);
196 if (s->first_pts != AV_NOPTS_VALUE) {
197 handle_trimming(ctx);
198 if (!avresample_available(s->avr))
199 return write_to_fifo(s, buf);
202 /* when we have two timestamps, compute how many samples would we have
203 * to add/remove to get proper sync between data and timestamps */
204 delta = pts - s->pts - get_delay(s);
205 out_size = avresample_available(s->avr);
207 if (llabs(delta) > s->min_delta ||
208 (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) {
209 av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
210 out_size = av_clipl_int32((int64_t)out_size + delta);
213 // adjust the compensation if delta is non-zero
214 int delay = get_delay(s);
215 int comp = s->comp + av_clip(delta * inlink->sample_rate / delay,
216 -s->max_comp, s->max_comp);
217 if (comp != s->comp) {
218 av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
219 if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) {
224 // adjust PTS to avoid monotonicity errors with input PTS jitter
230 AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size);
232 ret = AVERROR(ENOMEM);
236 if (s->first_frame && delta > 0) {
237 int planar = av_sample_fmt_is_planar(buf_out->format);
238 int planes = planar ? nb_channels : 1;
239 int block_size = av_get_bytes_per_sample(buf_out->format) *
240 (planar ? 1 : nb_channels);
244 av_samples_set_silence(buf_out->extended_data, 0, delta,
245 nb_channels, buf->format);
247 for (ch = 0; ch < planes; ch++)
248 buf_out->extended_data[ch] += delta * block_size;
250 avresample_read(s->avr, buf_out->extended_data, out_size);
252 for (ch = 0; ch < planes; ch++)
253 buf_out->extended_data[ch] -= delta * block_size;
255 avresample_read(s->avr, buf_out->extended_data, out_size);
258 av_samples_set_silence(buf_out->extended_data, out_size - delta,
259 delta, nb_channels, buf->format);
262 buf_out->pts = s->pts;
263 ret = ff_filter_frame(outlink, buf_out);
267 } else if (avresample_available(s->avr)) {
268 av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
272 /* drain any remaining buffered data */
273 avresample_read(s->avr, NULL, avresample_available(s->avr));
275 new_pts = pts - avresample_get_delay(s->avr);
276 /* check for s->pts monotonicity */
277 if (new_pts > s->pts) {
279 ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data,
280 buf->linesize[0], buf->nb_samples);
282 av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
294 static const AVFilterPad avfilter_af_asyncts_inputs[] = {
297 .type = AVMEDIA_TYPE_AUDIO,
298 .filter_frame = filter_frame
303 static const AVFilterPad avfilter_af_asyncts_outputs[] = {
306 .type = AVMEDIA_TYPE_AUDIO,
307 .config_props = config_props,
308 .request_frame = request_frame
313 AVFilter ff_af_asyncts = {
315 .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),
318 .priv_size = sizeof(ASyncContext),
319 .priv_class = &asyncts_class,
320 .inputs = avfilter_af_asyncts_inputs,
321 .outputs = avfilter_af_asyncts_outputs,