aboutsummaryrefslogtreecommitdiffstats
path: root/libavfilter/af_apad.c
blob: f7a4199c6486c8d81f33e1c1073be4ce53357b15 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
/*
 * Copyright (c) 2012 Michael Niedermayer
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

/**
 * @file
 * audio pad filter.
 *
 * Based on af_aresample.c
 */

#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/opt.h"
#include "libavutil/samplefmt.h"
#include "libavutil/avassert.h"
#include "avfilter.h"
#include "audio.h"
#include "internal.h"

typedef struct APadContext {
    const AVClass *class;
    int64_t next_pts;

    int packet_size;
    int64_t pad_len, pad_len_left;
    int64_t whole_len, whole_len_left;
} APadContext;

#define OFFSET(x) offsetof(APadContext, x)
#define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM

static const AVOption apad_options[] = {
    { "packet_size", "set silence packet size",                                  OFFSET(packet_size), AV_OPT_TYPE_INT,   { .i64 = 4096 }, 0, INT_MAX, A },
    { "pad_len",     "set number of samples of silence to add",                  OFFSET(pad_len),     AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A },
    { "whole_len",   "set minimum target number of samples in the audio stream", OFFSET(whole_len),   AV_OPT_TYPE_INT64, { .i64 = -1 }, -1, INT64_MAX, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(apad);

static av_cold int init(AVFilterContext *ctx)
{
    APadContext *s = ctx->priv;

    s->next_pts = AV_NOPTS_VALUE;
    if (s->whole_len >= 0 && s->pad_len >= 0) {
        av_log(ctx, AV_LOG_ERROR, "Both whole and pad length are set, this is not possible\n");
        return AVERROR(EINVAL);
    }
    s->pad_len_left   = s->pad_len;
    s->whole_len_left = s->whole_len;

    return 0;
}

static int filter_frame(AVFilterLink *inlink, AVFrame *frame)
{
    AVFilterContext *ctx = inlink->dst;
    APadContext *s = ctx->priv;

    if (s->whole_len >= 0) {
        s->whole_len_left = FFMAX(s->whole_len_left - frame->nb_samples, 0);
        av_log(ctx, AV_LOG_DEBUG,
               "n_out:%d whole_len_left:%"PRId64"\n", frame->nb_samples, s->whole_len_left);
    }

    s->next_pts = frame->pts + av_rescale_q(frame->nb_samples, (AVRational){1, inlink->sample_rate}, inlink->time_base);
    return ff_filter_frame(ctx->outputs[0], frame);
}

static int request_frame(AVFilterLink *outlink)
{
    AVFilterContext *ctx = outlink->src;
    APadContext *s = ctx->priv;
    int ret;

    ret = ff_request_frame(ctx->inputs[0]);

    if (ret == AVERROR_EOF && !ctx->is_disabled) {
        int n_out = s->packet_size;
        AVFrame *outsamplesref;

        if (s->whole_len >= 0 && s->pad_len < 0) {
            s->pad_len = s->pad_len_left = s->whole_len_left;
        }
        if (s->pad_len >=0 || s->whole_len >= 0) {
            n_out = FFMIN(n_out, s->pad_len_left);
            s->pad_len_left -= n_out;
            av_log(ctx, AV_LOG_DEBUG,
                   "padding n_out:%d pad_len_left:%"PRId64"\n", n_out, s->pad_len_left);
        }

        if (!n_out)
            return AVERROR_EOF;

        outsamplesref = ff_get_audio_buffer(outlink, n_out);
        if (!outsamplesref)
            return AVERROR(ENOMEM);

        av_assert0(outsamplesref->sample_rate == outlink->sample_rate);
        av_assert0(outsamplesref->nb_samples  == n_out);

        av_samples_set_silence(outsamplesref->extended_data, 0,
                               n_out,
                               outsamplesref->channels,
                               outsamplesref->format);

        outsamplesref->pts = s->next_pts;
        if (s->next_pts != AV_NOPTS_VALUE)
            s->next_pts += av_rescale_q(n_out, (AVRational){1, outlink->sample_rate}, outlink->time_base);

        return ff_filter_frame(outlink, outsamplesref);
    }
    return ret;
}

static const AVFilterPad apad_inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .filter_frame = filter_frame,
    },
    { NULL }
};

static const AVFilterPad apad_outputs[] = {
    {
        .name          = "default",
        .request_frame = request_frame,
        .type          = AVMEDIA_TYPE_AUDIO,
    },
    { NULL }
};

AVFilter ff_af_apad = {
    .name          = "apad",
    .description   = NULL_IF_CONFIG_SMALL("Pad audio with silence."),
    .init          = init,
    .priv_size     = sizeof(APadContext),
    .inputs        = apad_inputs,
    .outputs       = apad_outputs,
    .priv_class    = &apad_class,
    .flags         = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
};