FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_earwax.c
Date: 2022-12-09 07:38:14
Exec Total Coverage
Lines: 70 77 90.9%
Functions: 7 7 100.0%
Branches: 21 30 70.0%

Line Branch Exec Source
1 /*
2 * Copyright (c) 2011 Mina Nagy Zaki
3 * Copyright (c) 2000 Edward Beingessner And Sundry Contributors.
4 * This source code is freely redistributable and may be used for any purpose.
5 * This copyright notice must be maintained. Edward Beingessner And Sundry
6 * Contributors are not responsible for the consequences of using this
7 * software.
8 *
9 * This file is part of FFmpeg.
10 *
11 * FFmpeg is free software; you can redistribute it and/or
12 * modify it under the terms of the GNU Lesser General Public
13 * License as published by the Free Software Foundation; either
14 * version 2.1 of the License, or (at your option) any later version.
15 *
16 * FFmpeg is distributed in the hope that it will be useful,
17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
19 * Lesser General Public License for more details.
20 *
21 * You should have received a copy of the GNU Lesser General Public
22 * License along with FFmpeg; if not, write to the Free Software
23 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
24 */
25
26 /**
27 * @file
28 * Stereo Widening Effect. Adds audio cues to move stereo image in
29 * front of the listener. Adapted from the libsox earwax effect.
30 */
31
32 #include "libavutil/channel_layout.h"
33 #include "avfilter.h"
34 #include "audio.h"
35 #include "formats.h"
36
37 #define NUMTAPS 32
38
39 static const int8_t filt[NUMTAPS * 2] = {
40 /* 30° 330° */
41 4, -6, /* 32 tap stereo FIR filter. */
42 4, -11, /* One side filters as if the */
43 -1, -5, /* signal was from 30 degrees */
44 3, 3, /* from the ear, the other as */
45 -2, 5, /* if 330 degrees. */
46 -5, 0,
47 9, 1,
48 6, 3, /* Input */
49 -4, -1, /* Left Right */
50 -5, -3, /* __________ __________ */
51 -2, -5, /* | | | | */
52 -7, 1, /* .---| Hh,0(f) | | Hh,0(f) |---. */
53 6, -7, /* / |__________| |__________| \ */
54 30, -29, /* / \ / \ */
55 12, -3, /* / X \ */
56 -11, 4, /* / / \ \ */
57 -3, 7, /* ____V_____ __________V V__________ _____V____ */
58 -20, 23, /* | | | | | | | | */
59 2, 0, /* | Hh,30(f) | | Hh,330(f)| | Hh,330(f)| | Hh,30(f) | */
60 1, -6, /* |__________| |__________| |__________| |__________| */
61 -14, -5, /* \ ___ / \ ___ / */
62 15, -18, /* \ / \ / _____ \ / \ / */
63 6, 7, /* `->| + |<--' / \ `-->| + |<-' */
64 15, -10, /* \___/ _/ \_ \___/ */
65 -14, 22, /* \ / \ / \ / */
66 -7, -2, /* `--->| | | |<---' */
67 -4, 9, /* \_/ \_/ */
68 6, -12, /* */
69 6, -6, /* Headphones */
70 0, -11,
71 0, -5,
72 4, 0};
73
74 typedef struct EarwaxContext {
75 int16_t filter[2][NUMTAPS];
76 int16_t taps[4][NUMTAPS * 2];
77
78 AVFrame *frame[2];
79 } EarwaxContext;
80
81 1 static int query_formats(AVFilterContext *ctx)
82 {
83 static const int sample_rates[] = { 44100, -1 };
84 int ret;
85
86 1 AVFilterFormats *formats = NULL;
87 1 AVFilterChannelLayouts *layout = NULL;
88
89
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2 if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_S16P )) < 0 ||
90
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2 (ret = ff_set_common_formats (ctx , formats )) < 0 ||
91
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2 (ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
92
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2 (ret = ff_set_common_channel_layouts (ctx , layout )) < 0 ||
93 1 (ret = ff_set_common_samplerates_from_list(ctx, sample_rates)) < 0)
94 return ret;
95
96 1 return 0;
97 }
98
99 //FIXME: replace with DSPContext.scalarproduct_int16
100 160 static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin,
101 const int16_t *filt, int16_t *out)
102 {
103 int32_t sample;
104 int16_t j;
105
106
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82080 while (in < endin) {
107 81920 sample = 0;
108
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2703360 for (j = 0; j < NUMTAPS; j++)
109 2621440 sample += in[j] * filt[j];
110 81920 *out = av_clip_int16(sample >> 7);
111 81920 out++;
112 81920 in++;
113 }
114
115 160 return out;
116 }
117
118 1 static int config_input(AVFilterLink *inlink)
119 {
120 1 EarwaxContext *s = inlink->dst->priv;
121
122
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33 for (int i = 0; i < NUMTAPS; i++) {
123 32 s->filter[0][i] = filt[i * 2];
124 32 s->filter[1][i] = filt[i * 2 + 1];
125 }
126
127 1 return 0;
128 }
129
130 80 static void convolve(AVFilterContext *ctx, AVFrame *in,
131 int input_ch, int output_ch,
132 int filter_ch, int tap_ch)
133 {
134 80 EarwaxContext *s = ctx->priv;
135 int16_t *taps, *endin, *dst, *src;
136 int len;
137
138 80 taps = s->taps[tap_ch];
139 80 dst = (int16_t *)s->frame[input_ch]->data[output_ch];
140 80 src = (int16_t *)in->data[input_ch];
141
142 80 len = FFMIN(NUMTAPS, in->nb_samples);
143 // copy part of new input and process with saved input
144 80 memcpy(taps+NUMTAPS, src, len * sizeof(*taps));
145 80 dst = scalarproduct(taps, taps + len, s->filter[filter_ch], dst);
146
147 // process current input
148
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80 if (in->nb_samples >= NUMTAPS) {
149 80 endin = src + in->nb_samples - NUMTAPS;
150 80 scalarproduct(src, endin, s->filter[filter_ch], dst);
151
152 // save part of input for next round
153 80 memcpy(taps, endin, NUMTAPS * sizeof(*taps));
154 } else {
155 memmove(taps, taps + in->nb_samples, NUMTAPS * sizeof(*taps));
156 }
157 80 }
158
159 40 static void mix(AVFilterContext *ctx, AVFrame *out,
160 int output_ch, int f0, int f1, int i0, int i1)
161 {
162 40 EarwaxContext *s = ctx->priv;
163 40 const int16_t *srcl = (const int16_t *)s->frame[f0]->data[i0];
164 40 const int16_t *srcr = (const int16_t *)s->frame[f1]->data[i1];
165 40 int16_t *dst = (int16_t *)out->data[output_ch];
166
167
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41000 for (int n = 0; n < out->nb_samples; n++)
168 40960 dst[n] = av_clip_int16(srcl[n] + srcr[n]);
169 40 }
170
171 20 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
172 {
173 20 AVFilterContext *ctx = inlink->dst;
174 20 EarwaxContext *s = ctx->priv;
175 20 AVFilterLink *outlink = ctx->outputs[0];
176 20 AVFrame *out = ff_get_audio_buffer(outlink, in->nb_samples);
177
178
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60 for (int ch = 0; ch < 2; ch++) {
179
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40 if (!s->frame[ch] || s->frame[ch]->nb_samples < in->nb_samples) {
180 2 av_frame_free(&s->frame[ch]);
181 2 s->frame[ch] = ff_get_audio_buffer(outlink, in->nb_samples);
182
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2 if (!s->frame[ch]) {
183 av_frame_free(&in);
184 av_frame_free(&out);
185 return AVERROR(ENOMEM);
186 }
187 }
188 }
189
190
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20 if (!out) {
191 av_frame_free(&in);
192 return AVERROR(ENOMEM);
193 }
194 20 av_frame_copy_props(out, in);
195
196 20 convolve(ctx, in, 0, 0, 0, 0);
197 20 convolve(ctx, in, 0, 1, 1, 1);
198 20 convolve(ctx, in, 1, 0, 0, 2);
199 20 convolve(ctx, in, 1, 1, 1, 3);
200
201 20 mix(ctx, out, 0, 0, 1, 1, 0);
202 20 mix(ctx, out, 1, 0, 1, 0, 1);
203
204 20 av_frame_free(&in);
205 20 return ff_filter_frame(outlink, out);
206 }
207
208 1 static av_cold void uninit(AVFilterContext *ctx)
209 {
210 1 EarwaxContext *s = ctx->priv;
211
212 1 av_frame_free(&s->frame[0]);
213 1 av_frame_free(&s->frame[1]);
214 1 }
215
216 static const AVFilterPad earwax_inputs[] = {
217 {
218 .name = "default",
219 .type = AVMEDIA_TYPE_AUDIO,
220 .filter_frame = filter_frame,
221 .config_props = config_input,
222 },
223 };
224
225 static const AVFilterPad earwax_outputs[] = {
226 {
227 .name = "default",
228 .type = AVMEDIA_TYPE_AUDIO,
229 },
230 };
231
232 const AVFilter ff_af_earwax = {
233 .name = "earwax",
234 .description = NULL_IF_CONFIG_SMALL("Widen the stereo image."),
235 .priv_size = sizeof(EarwaxContext),
236 .uninit = uninit,
237 FILTER_INPUTS(earwax_inputs),
238 FILTER_OUTPUTS(earwax_outputs),
239 FILTER_QUERY_FUNC(query_formats),
240 };
241