FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_haas.c
Date: 2022-12-05 03:11:11
Exec Total Coverage
Lines: 0 74 0.0%
Functions: 0 4 0.0%
Branches: 0 31 0.0%

Line Branch Exec Source
1 /*
2 * Copyright (c) 2001-2010 Vladimir Sadovnikov
3 *
4 * This file is part of FFmpeg.
5 *
6 * FFmpeg is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
10 *
11 * FFmpeg is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
15 *
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with FFmpeg; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
19 */
20
21 #include "libavutil/channel_layout.h"
22 #include "libavutil/opt.h"
23 #include "avfilter.h"
24 #include "audio.h"
25 #include "formats.h"
26
27 #define MAX_HAAS_DELAY 40
28
29 typedef struct HaasContext {
30 const AVClass *class;
31
32 int par_m_source;
33 double par_delay0;
34 double par_delay1;
35 int par_phase0;
36 int par_phase1;
37 int par_middle_phase;
38 double par_side_gain;
39 double par_gain0;
40 double par_gain1;
41 double par_balance0;
42 double par_balance1;
43 double level_in;
44 double level_out;
45
46 double *buffer;
47 size_t buffer_size;
48 uint32_t write_ptr;
49 uint32_t delay[2];
50 double balance_l[2];
51 double balance_r[2];
52 double phase0;
53 double phase1;
54 } HaasContext;
55
56 #define OFFSET(x) offsetof(HaasContext, x)
57 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
58
59 static const AVOption haas_options[] = {
60 { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
61 { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
62 { "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
63 { "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, "source" },
64 { "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, "source" },
65 { "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, "source" },
66 { "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, "source" },
67 { "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, "source" },
68 { "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
69 { "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A },
70 { "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A },
71 { "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
72 { "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
73 { "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A },
74 { "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A },
75 { "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
76 { "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A },
77 { NULL }
78 };
79
80 AVFILTER_DEFINE_CLASS(haas);
81
82 static int query_formats(AVFilterContext *ctx)
83 {
84 AVFilterFormats *formats = NULL;
85 AVFilterChannelLayouts *layout = NULL;
86 int ret;
87
88 if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
89 (ret = ff_set_common_formats (ctx , formats )) < 0 ||
90 (ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
91 (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
92 return ret;
93
94 return ff_set_common_all_samplerates(ctx);
95 }
96
97 static int config_input(AVFilterLink *inlink)
98 {
99 AVFilterContext *ctx = inlink->dst;
100 HaasContext *s = ctx->priv;
101 size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001);
102 size_t new_buf_size = 1;
103
104 while (new_buf_size < min_buf_size)
105 new_buf_size <<= 1;
106
107 av_freep(&s->buffer);
108 s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
109 if (!s->buffer)
110 return AVERROR(ENOMEM);
111
112 s->buffer_size = new_buf_size;
113 s->write_ptr = 0;
114
115 s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
116 s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);
117
118 s->phase0 = s->par_phase0 ? 1.0 : -1.0;
119 s->phase1 = s->par_phase1 ? 1.0 : -1.0;
120
121 s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
122 s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
123 s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
124 s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;
125
126 return 0;
127 }
128
129 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
130 {
131 AVFilterContext *ctx = inlink->dst;
132 AVFilterLink *outlink = ctx->outputs[0];
133 HaasContext *s = ctx->priv;
134 const double *src = (const double *)in->data[0];
135 const double level_in = s->level_in;
136 const double level_out = s->level_out;
137 const uint32_t mask = s->buffer_size - 1;
138 double *buffer = s->buffer;
139 AVFrame *out;
140 double *dst;
141 int n;
142
143 if (av_frame_is_writable(in)) {
144 out = in;
145 } else {
146 out = ff_get_audio_buffer(outlink, in->nb_samples);
147 if (!out) {
148 av_frame_free(&in);
149 return AVERROR(ENOMEM);
150 }
151 av_frame_copy_props(out, in);
152 }
153 dst = (double *)out->data[0];
154
155 for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
156 double mid, side[2], side_l, side_r;
157 uint32_t s0_ptr, s1_ptr;
158
159 switch (s->par_m_source) {
160 case 0: mid = src[0]; break;
161 case 1: mid = src[1]; break;
162 case 2: mid = (src[0] + src[1]) * 0.5; break;
163 case 3: mid = (src[0] - src[1]) * 0.5; break;
164 }
165
166 mid *= level_in;
167
168 buffer[s->write_ptr] = mid;
169
170 s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
171 s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;
172
173 if (s->par_middle_phase)
174 mid = -mid;
175
176 side[0] = buffer[s0_ptr] * s->par_side_gain;
177 side[1] = buffer[s1_ptr] * s->par_side_gain;
178 side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
179 side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];
180
181 dst[0] = (mid + side_l) * level_out;
182 dst[1] = (mid + side_r) * level_out;
183
184 s->write_ptr = (s->write_ptr + 1) & mask;
185 }
186
187 if (out != in)
188 av_frame_free(&in);
189 return ff_filter_frame(outlink, out);
190 }
191
192 static av_cold void uninit(AVFilterContext *ctx)
193 {
194 HaasContext *s = ctx->priv;
195
196 av_freep(&s->buffer);
197 s->buffer_size = 0;
198 }
199
200 static const AVFilterPad inputs[] = {
201 {
202 .name = "default",
203 .type = AVMEDIA_TYPE_AUDIO,
204 .filter_frame = filter_frame,
205 .config_props = config_input,
206 },
207 };
208
209 static const AVFilterPad outputs[] = {
210 {
211 .name = "default",
212 .type = AVMEDIA_TYPE_AUDIO,
213 },
214 };
215
216 const AVFilter ff_af_haas = {
217 .name = "haas",
218 .description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
219 .priv_size = sizeof(HaasContext),
220 .priv_class = &haas_class,
221 .uninit = uninit,
222 FILTER_INPUTS(inputs),
223 FILTER_OUTPUTS(outputs),
224 FILTER_QUERY_FUNC(query_formats),
225 };
226