path: root/libavfilter/af_asetrate.c (plain)
blob: 66febd71ba6e9ad42627f2f32ef80201f6570f12

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;
27	int sample_rate;
28	int rescale_pts;
29	} ASetRateContext;
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
49	static av_cold int query_formats(AVFilterContext *ctx)
50	{
51	ASetRateContext *sr = ctx->priv;
52	int sample_rates[] = { sr->sample_rate, -1 };
53
54	return ff_formats_ref(ff_make_format_list(sample_rates),
55	&ctx->outputs[0]->in_samplerates);
56	}
57
58	static av_cold int config_props(AVFilterLink *outlink)
59	{
60	AVFilterContext *ctx = outlink->src;
61	ASetRateContext *sr = ctx->priv;
62	AVFilterLink *inlink = ctx->inputs[0];
63	AVRational intb = ctx->inputs[0]->time_base;
64	int inrate = inlink->sample_rate;
65
66	if (intb.num == 1 && intb.den == inrate) {
67	outlink->time_base.num = 1;
68	outlink->time_base.den = outlink->sample_rate;
69	} else {
70	outlink->time_base = intb;
71	sr->rescale_pts = 1;
72	if (av_q2d(intb) > 1.0 / FFMAX(inrate, outlink->sample_rate))
73	av_log(ctx, AV_LOG_WARNING, "Time base is inaccurate\n");
74	}
75	return 0;
76	}
77
78	static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
79	{
80	AVFilterContext *ctx = inlink->dst;
81	ASetRateContext *sr = ctx->priv;
82	AVFilterLink *outlink = ctx->outputs[0];
83
84	frame->sample_rate = outlink->sample_rate;
85	if (sr->rescale_pts)
86	frame->pts = av_rescale(frame->pts, inlink->sample_rate,
87	outlink->sample_rate);
88	return ff_filter_frame(outlink, frame);
89	}
90
91	static const AVFilterPad asetrate_inputs[] = {
92	{
93	.name = "default",
94	.type = AVMEDIA_TYPE_AUDIO,
95	.filter_frame = filter_frame,
96	},
97	{ NULL }
98	};
99
100	static const AVFilterPad asetrate_outputs[] = {
101	{
102	.name = "default",
103	.type = AVMEDIA_TYPE_AUDIO,
104	.config_props = config_props,
105	},
106	{ NULL }
107	};
108
109	AVFilter ff_af_asetrate = {
110	.name = "asetrate",
111	.description = NULL_IF_CONFIG_SMALL("Change the sample rate without "
112	"altering the data."),
113	.query_formats = query_formats,
114	.priv_size = sizeof(ASetRateContext),
115	.inputs = asetrate_inputs,
116	.outputs = asetrate_outputs,
117	.priv_class = &asetrate_class,
118	};
119