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
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
|
/*
* Copyright (c) 2011 Mina Nagy Zaki
* Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
* This source code is freely redistributable and may be used for any purpose.
* This copyright notice must be maintained. Edward Beingessner And Sundry
* Contributors are not responsible for the consequences of using this
* software.
*
* 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
* Stereo Widening Effect. Adds audio cues to move stereo image in
* front of the listener. Adapted from the libsox earwax effect.
*/
#include "libavutil/channel_layout.h"
#include "avfilter.h"
#include "audio.h"
#include "formats.h"
#define NUMTAPS 32
static const int8_t filt[NUMTAPS * 2] = {
/* 30° 330° */
4, -6, /* 32 tap stereo FIR filter. */
4, -11, /* One side filters as if the */
-1, -5, /* signal was from 30 degrees */
3, 3, /* from the ear, the other as */
-2, 5, /* if 330 degrees. */
-5, 0,
9, 1,
6, 3, /* Input */
-4, -1, /* Left Right */
-5, -3, /* __________ __________ */
-2, -5, /* | | | | */
-7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
6, -7, /* / |__________| |__________| \ */
30, -29, /* / \ / \ */
12, -3, /* / X \ */
-11, 4, /* / / \ \ */
-3, 7, /* ____V_____ __________V V__________ _____V____ */
-20, 23, /* | | | | | | | | */
2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
1, -6, /* |__________| |__________| |__________| |__________| */
-14, -5, /* \ ___ / \ ___ / */
15, -18, /* \ / \ / _____ \ / \ / */
6, 7, /* `->| + |<--' / \ `-->| + |<-' */
15, -10, /* \___/ _/ \_ \___/ */
-14, 22, /* \ / \ / \ / */
-7, -2, /* `--->| | | |<---' */
-4, 9, /* \_/ \_/ */
6, -12, /* */
6, -6, /* Headphones */
0, -11,
0, -5,
4, 0};
typedef struct EarwaxContext {
int16_t filter[2][NUMTAPS];
int16_t taps[4][NUMTAPS * 2];
AVFrame *frame[2];
} EarwaxContext;
static int query_formats(AVFilterContext *ctx)
{
static const int sample_rates[] = { 44100, -1 };
int ret;
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layout = NULL;
if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
(ret = ff_set_common_formats (ctx , formats )) < 0 ||
(ret = ff_add_channel_layout (&layout , AV_CH_LAYOUT_STEREO )) < 0 ||
(ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
(ret = ff_set_common_samplerates (ctx , ff_make_format_list(sample_rates) )) < 0)
return ret;
return 0;
}
//FIXME: replace with DSPContext.scalarproduct_int16
static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
const int16_t *filt, int16_t *out)
{
int32_t sample;
int16_t j;
while (in < endin) {
sample = 0;
for (j = 0; j < NUMTAPS; j++)
sample += in[j] * filt[j];
*out = av_clip_int16(sample >> 7);
out++;
in++;
}
return out;
}
static int config_input(AVFilterLink *inlink)
{
EarwaxContext *s = inlink->dst->priv;
for (int i = 0; i < NUMTAPS; i++) {
s->filter[0][i] = filt[i * 2];
s->filter[1][i] = filt[i * 2 + 1];
}
return 0;
}
static void convolve(AVFilterContext *ctx, AVFrame *in,
int input_ch, int output_ch,
int filter_ch, int tap_ch)
{
EarwaxContext *s = ctx->priv;
int16_t *taps, *endin, *dst, *src;
int len;
taps = s->taps[tap_ch];
dst = (int16_t *)s->frame[input_ch]->data[output_ch];
src = (int16_t *)in->data[input_ch];
len = FFMIN(NUMTAPS, in->nb_samples);
// copy part of new input and process with saved input
memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
// process current input
if (in->nb_samples >= NUMTAPS) {
endin = src + in->nb_samples - NUMTAPS;
scalarproduct(src, endin, s->filter[filter_ch], dst);
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
} else {
memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
}
}
static void mix(AVFilterContext *ctx, AVFrame *out,
int output_ch, int f0, int f1, int i0, int i1)
{
EarwaxContext *s = ctx->priv;
const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
int16_t *dst = (int16_t *)out->data[output_ch];
for (int n = 0; n < out->nb_samples; n++)
dst[n] = av_clip_int16(srcl[n] + srcr[n]);
}
static int filter_frame(AVFilterLink *inlink, AVFrame *in)
{
AVFilterContext *ctx = inlink->dst;
EarwaxContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
for (int ch = 0; ch < 2; ch++) {
if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
av_frame_free(&s->frame[ch]);
s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
if (!s->frame[ch]) {
av_frame_free(&in);
av_frame_free(&out);
return AVERROR(ENOMEM);
}
}
}
if (!out) {
av_frame_free(&in);
return AVERROR(ENOMEM);
}
av_frame_copy_props(out, in);
convolve(ctx, in, 0, 0, 0, 0);
convolve(ctx, in, 0, 1, 1, 1);
convolve(ctx, in, 1, 0, 0, 2);
convolve(ctx, in, 1, 1, 1, 3);
mix(ctx, out, 0, 0, 1, 1, 0);
mix(ctx, out, 1, 0, 1, 0, 1);
av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
static av_cold void uninit(AVFilterContext *ctx)
{
EarwaxContext *s = ctx->priv;
av_frame_free(&s->frame[0]);
av_frame_free(&s->frame[1]);
}
static const AVFilterPad earwax_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_frame = filter_frame,
.config_props = config_input,
},
{ NULL }
};
static const AVFilterPad earwax_outputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
},
{ NULL }
};
const AVFilter ff_af_earwax = {
.name = "earwax",
.description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
.query_formats = query_formats,
.priv_size = sizeof(EarwaxContext),
.uninit = uninit,
.inputs = earwax_inputs,
.outputs = earwax_outputs,
};
|
