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
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
|
/*
* Copyright (c) 2011 Stefano Sabatini
*
* 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 volume filter
* based on ffmpeg.c code
*/
#include "libavutil/audioconvert.h"
#include "libavutil/eval.h"
#include "audio.h"
#include "avfilter.h"
typedef struct {
double volume;
int volume_i;
} VolumeContext;
static av_cold int init(AVFilterContext *ctx, const char *args, void *opaque)
{
VolumeContext *vol = ctx->priv;
char *tail;
int ret = 0;
vol->volume = 1.0;
if (args) {
/* parse the number as a decimal number */
double d = strtod(args, &tail);
if (*tail) {
if (!strcmp(tail, "dB")) {
/* consider the argument an adjustement in decibels */
d = pow(10, d/20);
} else {
/* parse the argument as an expression */
ret = av_expr_parse_and_eval(&d, args, NULL, NULL,
NULL, NULL, NULL, NULL,
NULL, 0, ctx);
}
}
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR,
"Invalid volume argument '%s'\n", args);
return AVERROR(EINVAL);
}
if (d < 0 || d > 65536) { /* 65536 = INT_MIN / (128 * 256) */
av_log(ctx, AV_LOG_ERROR,
"Negative or too big volume value %f\n", d);
return AVERROR(EINVAL);
}
vol->volume = d;
}
vol->volume_i = (int)(vol->volume * 256 + 0.5);
av_log(ctx, AV_LOG_INFO, "volume=%f\n", vol->volume);
return 0;
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterFormats *formats = NULL;
enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_U8,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_S32,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_DBL,
AV_SAMPLE_FMT_NONE
};
int packing_fmts[] = { AVFILTER_PACKED, -1 };
formats = avfilter_make_all_channel_layouts();
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_channel_layouts(ctx, formats);
formats = avfilter_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_sample_formats(ctx, formats);
formats = avfilter_make_format_list(packing_fmts);
if (!formats)
return AVERROR(ENOMEM);
avfilter_set_common_packing_formats(ctx, formats);
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
VolumeContext *vol = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
const int nb_samples = insamples->audio->nb_samples *
av_get_channel_layout_nb_channels(insamples->audio->channel_layout);
const double volume = vol->volume;
const int volume_i = vol->volume_i;
int i;
if (volume_i != 256) {
switch (insamples->format) {
case AV_SAMPLE_FMT_U8:
{
uint8_t *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
int v = (((*p - 128) * volume_i + 128) >> 8) + 128;
*p++ = av_clip_uint8(v);
}
break;
}
case AV_SAMPLE_FMT_S16:
{
int16_t *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
int v = ((int64_t)*p * volume_i + 128) >> 8;
*p++ = av_clip_int16(v);
}
break;
}
case AV_SAMPLE_FMT_S32:
{
int32_t *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
int64_t v = (((int64_t)*p * volume_i + 128) >> 8);
*p++ = av_clipl_int32(v);
}
break;
}
case AV_SAMPLE_FMT_FLT:
{
float *p = (void *)insamples->data[0];
float scale = (float)volume;
for (i = 0; i < nb_samples; i++) {
*p++ *= scale;
}
break;
}
case AV_SAMPLE_FMT_DBL:
{
double *p = (void *)insamples->data[0];
for (i = 0; i < nb_samples; i++) {
*p *= volume;
p++;
}
break;
}
}
}
ff_filter_samples(outlink, insamples);
}
AVFilter avfilter_af_volume = {
.name = "volume",
.description = NULL_IF_CONFIG_SMALL("Change input volume."),
.query_formats = query_formats,
.priv_size = sizeof(VolumeContext),
.init = init,
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ|AV_PERM_WRITE},
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO, },
{ .name = NULL}},
};
|