2 * This file is part of Libav.
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.
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.
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
19 #include "libavresample/avresample.h"
20 #include "libavutil/audio_fifo.h"
21 #include "libavutil/mathematics.h"
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
29 typedef struct ASyncContext {
32 AVAudioResampleContext *avr;
33 int64_t pts; ///< timestamp in samples of the first sample in fifo
34 int min_delta; ///< pad/trim min threshold in samples
41 /* set by filter_samples() to signal an output frame to request_frame() */
45 #define OFFSET(x) offsetof(ASyncContext, x)
46 #define A AV_OPT_FLAG_AUDIO_PARAM
47 static const AVOption asyncts_options[] = {
48 { "compensate", "Stretch/squeeze the data to make it match the timestamps", OFFSET(resample), AV_OPT_TYPE_INT, { 0 }, 0, 1, A },
49 { "min_delta", "Minimum difference between timestamps and audio data "
50 "(in seconds) to trigger padding/trimmin the data.", OFFSET(min_delta_sec), AV_OPT_TYPE_FLOAT, { 0.1 }, 0, INT_MAX, A },
51 { "max_comp", "Maximum compensation in samples per second.", OFFSET(max_comp), AV_OPT_TYPE_INT, { 500 }, 0, INT_MAX, A },
55 AVFILTER_DEFINE_CLASS(asyncts);
57 static int init(AVFilterContext *ctx, const char *args)
59 ASyncContext *s = ctx->priv;
62 s->class = &asyncts_class;
63 av_opt_set_defaults(s);
65 if ((ret = av_set_options_string(s, args, "=", ":")) < 0) {
66 av_log(ctx, AV_LOG_ERROR, "Error parsing options string '%s'.\n", args);
71 s->pts = AV_NOPTS_VALUE;
76 static 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 static int request_frame(AVFilterLink *link)
116 AVFilterContext *ctx = link->src;
117 ASyncContext *s = ctx->priv;
122 while (ret >= 0 && !s->got_output)
123 ret = ff_request_frame(ctx->inputs[0]);
126 if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
127 AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
130 return AVERROR(ENOMEM);
131 avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
132 nb_samples, NULL, 0, 0);
134 return ff_filter_samples(link, buf);
140 static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
142 int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
143 buf->linesize[0], buf->audio->nb_samples);
144 avfilter_unref_buffer(buf);
148 /* get amount of data currently buffered, in samples */
149 static int64_t get_delay(ASyncContext *s)
151 return avresample_available(s->avr) + avresample_get_delay(s->avr);
154 static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
156 AVFilterContext *ctx = inlink->dst;
157 ASyncContext *s = ctx->priv;
158 AVFilterLink *outlink = ctx->outputs[0];
159 int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
160 int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
161 av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
165 /* buffer data until we get the first timestamp */
166 if (s->pts == AV_NOPTS_VALUE) {
167 if (pts != AV_NOPTS_VALUE) {
168 s->pts = pts - get_delay(s);
170 return write_to_fifo(s, buf);
173 /* now wait for the next timestamp */
174 if (pts == AV_NOPTS_VALUE) {
175 return write_to_fifo(s, buf);
178 /* when we have two timestamps, compute how many samples would we have
179 * to add/remove to get proper sync between data and timestamps */
180 delta = pts - s->pts - get_delay(s);
181 out_size = avresample_available(s->avr);
183 if (labs(delta) > s->min_delta) {
184 av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta);
188 int comp = av_clip(delta, -s->max_comp, s->max_comp);
189 av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp);
190 avresample_set_compensation(s->avr, delta, inlink->sample_rate);
196 AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
199 ret = AVERROR(ENOMEM);
203 avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
204 buf_out->pts = s->pts;
207 av_samples_set_silence(buf_out->extended_data, out_size - delta,
208 delta, nb_channels, buf->format);
210 ret = ff_filter_samples(outlink, buf_out);
215 av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
219 /* drain any remaining buffered data */
220 avresample_read(s->avr, NULL, avresample_available(s->avr));
222 s->pts = pts - avresample_get_delay(s->avr);
223 ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
224 buf->linesize[0], buf->audio->nb_samples);
227 avfilter_unref_buffer(buf);
232 AVFilter avfilter_af_asyncts = {
234 .description = NULL_IF_CONFIG_SMALL("Sync audio data to timestamps"),
239 .priv_size = sizeof(ASyncContext),
241 .inputs = (const AVFilterPad[]) {{ .name = "default",
242 .type = AVMEDIA_TYPE_AUDIO,
243 .filter_samples = filter_samples },
245 .outputs = (const AVFilterPad[]) {{ .name = "default",
246 .type = AVMEDIA_TYPE_AUDIO,
247 .config_props = config_props,
248 .request_frame = request_frame },