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