aboutsummaryrefslogtreecommitdiffstats
path: root/libavfilter/af_dcshift.c
blob: a722928b184f19aae2a1749a3cf837534739ed55 (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
/*
 * Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
 * Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
 *
 * 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
 */

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

typedef struct DCShiftContext {
    const AVClass *class;
    double dcshift;
    double limiterthreshold;
    double limitergain;
} DCShiftContext;

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

static const AVOption dcshift_options[] = {
    { "shift",       "set DC shift",     OFFSET(dcshift),       AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
    { "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
    { NULL }
};

AVFILTER_DEFINE_CLASS(dcshift);

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

    s->limiterthreshold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));

    return 0;
}

static int query_formats(AVFilterContext *ctx)
{
    static const enum AVSampleFormat sample_fmts[] = {
        AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
    };
    int ret = ff_set_common_all_channel_counts(ctx);
    if (ret < 0)
        return ret;

    ret = ff_set_common_formats_from_list(ctx, sample_fmts);
    if (ret < 0)
        return ret;

    return ff_set_common_all_samplerates(ctx);
}

static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
    AVFilterContext *ctx = inlink->dst;
    AVFilterLink *outlink = ctx->outputs[0];
    AVFrame *out;
    DCShiftContext *s = ctx->priv;
    int i, j;
    double dcshift = s->dcshift;

    if (av_frame_is_writable(in)) {
        out = in;
    } else {
        out = ff_get_audio_buffer(outlink, in->nb_samples);
        if (!out) {
            av_frame_free(&in);
            return AVERROR(ENOMEM);
        }
        av_frame_copy_props(out, in);
    }

    if (s->limitergain > 0) {
        for (i = 0; i < inlink->channels; i++) {
            const int32_t *src = (int32_t *)in->extended_data[i];
            int32_t *dst = (int32_t *)out->extended_data[i];

            for (j = 0; j < in->nb_samples; j++) {
                double d;

                d = src[j];

                if (d > s->limiterthreshold && dcshift > 0) {
                    d = (d - s->limiterthreshold) * s->limitergain /
                             (INT32_MAX - s->limiterthreshold) +
                             s->limiterthreshold + dcshift;
                } else if (d < -s->limiterthreshold && dcshift < 0) {
                    d = (d + s->limiterthreshold) * s->limitergain /
                             (INT32_MAX - s->limiterthreshold) -
                             s->limiterthreshold + dcshift;
                } else {
                    d = dcshift * INT32_MAX + d;
                }

                dst[j] = av_clipl_int32(d);
            }
        }
    } else {
        for (i = 0; i < inlink->channels; i++) {
            const int32_t *src = (int32_t *)in->extended_data[i];
            int32_t *dst = (int32_t *)out->extended_data[i];

            for (j = 0; j < in->nb_samples; j++) {
                double d = dcshift * (INT32_MAX + 1.) + src[j];

                dst[j] = av_clipl_int32(d);
            }
        }
    }

    if (out != in)
        av_frame_free(&in);
    return ff_filter_frame(outlink, out);
}
static const AVFilterPad dcshift_inputs[] = {
    {
        .name         = "default",
        .type         = AVMEDIA_TYPE_AUDIO,
        .filter_frame = filter_frame,
    },
    { NULL }
};

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

const AVFilter ff_af_dcshift = {
    .name           = "dcshift",
    .description    = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
    .query_formats  = query_formats,
    .priv_size      = sizeof(DCShiftContext),
    .priv_class     = &dcshift_class,
    .init           = init,
    .inputs         = dcshift_inputs,
    .outputs        = dcshift_outputs,
    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
};