1 /*
2 * Copyright (c) 2000 Chris Ausbrooks <weed@bucket.pp.ualr.edu>
3 * Copyright (c) 2000 Fabien COELHO <fabien@coelho.net>
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22 #include "libavutil/opt.h"
23 #include "libavutil/samplefmt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "internal.h"
27
28 typedef struct DCShiftContext {
29 const AVClass *class;
30 double dcshift;
31 double limiterthreshold;
32 double limitergain;
33 } DCShiftContext;
34
35 #define OFFSET(x) offsetof(DCShiftContext, x)
36 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
37
38 static const AVOption dcshift_options[] = {
39 { "shift", "set DC shift", OFFSET(dcshift), AV_OPT_TYPE_DOUBLE, {.dbl=0}, -1, 1, A },
40 { "limitergain", "set limiter gain", OFFSET(limitergain), AV_OPT_TYPE_DOUBLE, {.dbl=0}, 0, 1, A },
41 { NULL }
42 };
43
44 AVFILTER_DEFINE_CLASS(dcshift);
45
init(AVFilterContext * ctx)46 static av_cold int init(AVFilterContext *ctx)
47 {
48 DCShiftContext *s = ctx->priv;
49
50 s->limiterthreshold = INT32_MAX * (1.0 - (fabs(s->dcshift) - s->limitergain));
51
52 return 0;
53 }
54
query_formats(AVFilterContext * ctx)55 static int query_formats(AVFilterContext *ctx)
56 {
57 AVFilterChannelLayouts *layouts;
58 AVFilterFormats *formats;
59 static const enum AVSampleFormat sample_fmts[] = {
60 AV_SAMPLE_FMT_S32P, AV_SAMPLE_FMT_NONE
61 };
62 int ret;
63
64 layouts = ff_all_channel_counts();
65 if (!layouts)
66 return AVERROR(ENOMEM);
67 ret = ff_set_common_channel_layouts(ctx, layouts);
68 if (ret < 0)
69 return ret;
70
71 formats = ff_make_format_list(sample_fmts);
72 if (!formats)
73 return AVERROR(ENOMEM);
74 ret = ff_set_common_formats(ctx, formats);
75 if (ret < 0)
76 return ret;
77
78 formats = ff_all_samplerates();
79 if (!formats)
80 return AVERROR(ENOMEM);
81 return ff_set_common_samplerates(ctx, formats);
82 }
83
filter_frame(AVFilterLink * inlink,AVFrame * in)84 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
85 {
86 AVFilterContext *ctx = inlink->dst;
87 AVFilterLink *outlink = ctx->outputs[0];
88 AVFrame *out;
89 DCShiftContext *s = ctx->priv;
90 int i, j;
91 double dcshift = s->dcshift;
92
93 if (av_frame_is_writable(in)) {
94 out = in;
95 } else {
96 out = ff_get_audio_buffer(outlink, in->nb_samples);
97 if (!out) {
98 av_frame_free(&in);
99 return AVERROR(ENOMEM);
100 }
101 av_frame_copy_props(out, in);
102 }
103
104 if (s->limitergain > 0) {
105 for (i = 0; i < inlink->channels; i++) {
106 const int32_t *src = (int32_t *)in->extended_data[i];
107 int32_t *dst = (int32_t *)out->extended_data[i];
108
109 for (j = 0; j < in->nb_samples; j++) {
110 double d;
111
112 d = src[j];
113
114 if (d > s->limiterthreshold && dcshift > 0) {
115 d = (d - s->limiterthreshold) * s->limitergain /
116 (INT32_MAX - s->limiterthreshold) +
117 s->limiterthreshold + dcshift;
118 } else if (d < -s->limiterthreshold && dcshift < 0) {
119 d = (d + s->limiterthreshold) * s->limitergain /
120 (INT32_MAX - s->limiterthreshold) -
121 s->limiterthreshold + dcshift;
122 } else {
123 d = dcshift * INT32_MAX + d;
124 }
125
126 dst[j] = av_clipl_int32(d);
127 }
128 }
129 } else {
130 for (i = 0; i < inlink->channels; i++) {
131 const int32_t *src = (int32_t *)in->extended_data[i];
132 int32_t *dst = (int32_t *)out->extended_data[i];
133
134 for (j = 0; j < in->nb_samples; j++) {
135 double d = dcshift * (INT32_MAX + 1.) + src[j];
136
137 dst[j] = av_clipl_int32(d);
138 }
139 }
140 }
141
142 if (out != in)
143 av_frame_free(&in);
144 return ff_filter_frame(outlink, out);
145 }
146 static const AVFilterPad dcshift_inputs[] = {
147 {
148 .name = "default",
149 .type = AVMEDIA_TYPE_AUDIO,
150 .filter_frame = filter_frame,
151 },
152 { NULL }
153 };
154
155 static const AVFilterPad dcshift_outputs[] = {
156 {
157 .name = "default",
158 .type = AVMEDIA_TYPE_AUDIO,
159 },
160 { NULL }
161 };
162
163 AVFilter ff_af_dcshift = {
164 .name = "dcshift",
165 .description = NULL_IF_CONFIG_SMALL("Apply a DC shift to the audio."),
166 .query_formats = query_formats,
167 .priv_size = sizeof(DCShiftContext),
168 .priv_class = &dcshift_class,
169 .init = init,
170 .inputs = dcshift_inputs,
171 .outputs = dcshift_outputs,
172 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
173 };
174