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af_asetrate.c
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1 /*
2  * Copyright (c) 2013 Nicolas George
3  *
4  * This file is part of FFmpeg.
5  *
6  * FFmpeg is free software; you can redistribute it and/or
7  * modify it under the terms of the GNU Lesser General Public License
8  * as published by the Free Software Foundation; either
9  * version 2.1 of the License, or (at your option) any later version.
10  *
11  * FFmpeg is distributed in the hope that it will be useful,
12  * but WITHOUT ANY WARRANTY; without even the implied warranty of
13  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
14  * GNU Lesser General Public License for more details.
15  *
16  * You should have received a copy of the GNU Lesser General Public License
17  * along with FFmpeg; if not, write to the Free Software Foundation, Inc.,
18  * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19  */
20 
21 #include "libavutil/opt.h"
22 #include "avfilter.h"
23 #include "internal.h"
24 
25 typedef struct {
26  const AVClass *class;
30 
31 #define CONTEXT ASetRateContext
32 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
33 
34 #define OPT_GENERIC(name, field, def, min, max, descr, type, deffield, ...) \
35  { name, descr, offsetof(CONTEXT, field), AV_OPT_TYPE_ ## type, \
36  { .deffield = def }, min, max, FLAGS, __VA_ARGS__ }
37 
38 #define OPT_INT(name, field, def, min, max, descr, ...) \
39  OPT_GENERIC(name, field, def, min, max, descr, INT, i64, __VA_ARGS__)
40 
41 static const AVOption asetrate_options[] = {
42  OPT_INT("sample_rate", sample_rate, 44100, 1, INT_MAX, "set the sample rate"),
43  OPT_INT("r", sample_rate, 44100, 1, INT_MAX, "set the sample rate"),
44  {NULL},
45 };
46 
47 AVFILTER_DEFINE_CLASS(asetrate);
48 
50 {
51  ASetRateContext *sr = ctx->priv;
52  int sample_rates[] = { sr->sample_rate, -1 };
53 
54  ff_formats_ref(ff_make_format_list(sample_rates),
55  &ctx->outputs[0]->in_samplerates);
56  return 0;
57 }
58 
59 static av_cold int config_props(AVFilterLink *outlink)
60 {
61  AVFilterContext *ctx = outlink->src;
62  ASetRateContext *sr = ctx->priv;
63  AVFilterLink *inlink = ctx->inputs[0];
64  AVRational intb = ctx->inputs[0]->time_base;
65  int inrate = inlink->sample_rate;
66 
67  if (intb.num == 1 && intb.den == inrate) {
68  outlink->time_base.num = 1;
69  outlink->time_base.den = outlink->sample_rate;
70  } else {
71  outlink->time_base = intb;
72  sr->rescale_pts = 1;
73  if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
74  av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
75  }
76  return 0;
77 }
78 
79 static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
80 {
81  AVFilterContext *ctx = inlink->dst;
82  ASetRateContext *sr = ctx->priv;
83  AVFilterLink *outlink = ctx->outputs[0];
84 
85  frame->sample_rate = outlink->sample_rate;
86  if (sr->rescale_pts)
87  frame->pts = av_rescale(frame->pts, inlink->sample_rate,
88  outlink->sample_rate);
89  return ff_filter_frame(outlink, frame);
90 }
91 
92 static const AVFilterPad asetrate_inputs[] = {
93  {
94  .name = "default",
95  .type = AVMEDIA_TYPE_AUDIO,
96  .filter_frame = filter_frame,
97  },
98  { NULL }
99 };
100 
101 static const AVFilterPad asetrate_outputs[] = {
102  {
103  .name = "default",
104  .type = AVMEDIA_TYPE_AUDIO,
105  .config_props = config_props,
106  },
107  { NULL }
108 };
109 
111  .name = "asetrate",
112  .description = NULL_IF_CONFIG_SMALL("Change the sample rate without "
113  "altering the data."),
114  .query_formats = query_formats,
115  .priv_size = sizeof(ASetRateContext),
116  .inputs = asetrate_inputs,
117  .outputs = asetrate_outputs,
118  .priv_class = &asetrate_class,
119 };