FFmpeg coverage


Directory: ../../../ffmpeg/
File: src/libavfilter/af_haas.c
Date: 2024-04-12 08:31:17
Exec Total Coverage
Lines: 0 75 0.0%
Functions: 0 4 0.0%
Branches: 0 31 0.0%

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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/mem.h"
23 #include "libavutil/opt.h"
24 #include "avfilter.h"
25 #include "audio.h"
26 #include "formats.h"
27
28 #define MAX_HAAS_DELAY 40
29
30 typedef struct HaasContext {
31 const AVClass *class;
32
33 int par_m_source;
34 double par_delay0;
35 double par_delay1;
36 int par_phase0;
37 int par_phase1;
38 int par_middle_phase;
39 double par_side_gain;
40 double par_gain0;
41 double par_gain1;
42 double par_balance0;
43 double par_balance1;
44 double level_in;
45 double level_out;
46
47 double *buffer;
48 size_t buffer_size;
49 uint32_t write_ptr;
50 uint32_t delay[2];
51 double balance_l[2];
52 double balance_r[2];
53 double phase0;
54 double phase1;
55 } HaasContext;
56
57 #define OFFSET(x) offsetof(HaasContext, x)
58 #define A AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
59
60 static const AVOption haas_options[] = {
61 { "level_in", "set level in", OFFSET(level_in), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
62 { "level_out", "set level out", OFFSET(level_out), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
63 { "side_gain", "set side gain", OFFSET(par_side_gain), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
64 { "middle_source", "set middle source", OFFSET(par_m_source), AV_OPT_TYPE_INT, {.i64=2}, 0, 3, A, .unit = "source" },
65 { "left", 0, 0, AV_OPT_TYPE_CONST, {.i64=0}, 0, 0, A, .unit = "source" },
66 { "right", 0, 0, AV_OPT_TYPE_CONST, {.i64=1}, 0, 0, A, .unit = "source" },
67 { "mid", "L+R", 0, AV_OPT_TYPE_CONST, {.i64=2}, 0, 0, A, .unit = "source" },
68 { "side", "L-R", 0, AV_OPT_TYPE_CONST, {.i64=3}, 0, 0, A, .unit = "source" },
69 { "middle_phase", "set middle phase", OFFSET(par_middle_phase), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
70 { "left_delay", "set left delay", OFFSET(par_delay0), AV_OPT_TYPE_DOUBLE, {.dbl=2.05}, 0, MAX_HAAS_DELAY, A },
71 { "left_balance", "set left balance", OFFSET(par_balance0), AV_OPT_TYPE_DOUBLE, {.dbl=-1.0}, -1, 1, A },
72 { "left_gain", "set left gain", OFFSET(par_gain0), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
73 { "left_phase", "set left phase", OFFSET(par_phase0), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, A },
74 { "right_delay", "set right delay", OFFSET(par_delay1), AV_OPT_TYPE_DOUBLE, {.dbl=2.12}, 0, MAX_HAAS_DELAY, A },
75 { "right_balance", "set right balance", OFFSET(par_balance1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, -1, 1, A },
76 { "right_gain", "set right gain", OFFSET(par_gain1), AV_OPT_TYPE_DOUBLE, {.dbl=1}, 0.015625, 64, A },
77 { "right_phase", "set right phase", OFFSET(par_phase1), AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, A },
78 { NULL }
79 };
80
81 AVFILTER_DEFINE_CLASS(haas);
82
83 static int query_formats(AVFilterContext *ctx)
84 {
85 AVFilterFormats *formats = NULL;
86 AVFilterChannelLayouts *layout = NULL;
87 int ret;
88
89 if ((ret = ff_add_format (&formats, AV_SAMPLE_FMT_DBL )) < 0 ||
90 (ret = ff_set_common_formats (ctx , formats )) < 0 ||
91 (ret = ff_add_channel_layout (&layout , &(AVChannelLayout)AV_CHANNEL_LAYOUT_STEREO)) < 0 ||
92 (ret = ff_set_common_channel_layouts (ctx , layout )) < 0)
93 return ret;
94
95 return ff_set_common_all_samplerates(ctx);
96 }
97
98 static int config_input(AVFilterLink *inlink)
99 {
100 AVFilterContext *ctx = inlink->dst;
101 HaasContext *s = ctx->priv;
102 size_t min_buf_size = (size_t)(inlink->sample_rate * MAX_HAAS_DELAY * 0.001);
103 size_t new_buf_size = 1;
104
105 while (new_buf_size < min_buf_size)
106 new_buf_size <<= 1;
107
108 av_freep(&s->buffer);
109 s->buffer = av_calloc(new_buf_size, sizeof(*s->buffer));
110 if (!s->buffer)
111 return AVERROR(ENOMEM);
112
113 s->buffer_size = new_buf_size;
114 s->write_ptr = 0;
115
116 s->delay[0] = (uint32_t)(s->par_delay0 * 0.001 * inlink->sample_rate);
117 s->delay[1] = (uint32_t)(s->par_delay1 * 0.001 * inlink->sample_rate);
118
119 s->phase0 = s->par_phase0 ? 1.0 : -1.0;
120 s->phase1 = s->par_phase1 ? 1.0 : -1.0;
121
122 s->balance_l[0] = (s->par_balance0 + 1) / 2 * s->par_gain0 * s->phase0;
123 s->balance_r[0] = (1.0 - (s->par_balance0 + 1) / 2) * (s->par_gain0) * s->phase0;
124 s->balance_l[1] = (s->par_balance1 + 1) / 2 * s->par_gain1 * s->phase1;
125 s->balance_r[1] = (1.0 - (s->par_balance1 + 1) / 2) * (s->par_gain1) * s->phase1;
126
127 return 0;
128 }
129
130 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
131 {
132 AVFilterContext *ctx = inlink->dst;
133 AVFilterLink *outlink = ctx->outputs[0];
134 HaasContext *s = ctx->priv;
135 const double *src = (const double *)in->data[0];
136 const double level_in = s->level_in;
137 const double level_out = s->level_out;
138 const uint32_t mask = s->buffer_size - 1;
139 double *buffer = s->buffer;
140 AVFrame *out;
141 double *dst;
142 int n;
143
144 if (av_frame_is_writable(in)) {
145 out = in;
146 } else {
147 out = ff_get_audio_buffer(outlink, in->nb_samples);
148 if (!out) {
149 av_frame_free(&in);
150 return AVERROR(ENOMEM);
151 }
152 av_frame_copy_props(out, in);
153 }
154 dst = (double *)out->data[0];
155
156 for (n = 0; n < in->nb_samples; n++, src += 2, dst += 2) {
157 double mid, side[2], side_l, side_r;
158 uint32_t s0_ptr, s1_ptr;
159
160 switch (s->par_m_source) {
161 case 0: mid = src[0]; break;
162 case 1: mid = src[1]; break;
163 case 2: mid = (src[0] + src[1]) * 0.5; break;
164 case 3: mid = (src[0] - src[1]) * 0.5; break;
165 }
166
167 mid *= level_in;
168
169 buffer[s->write_ptr] = mid;
170
171 s0_ptr = (s->write_ptr + s->buffer_size - s->delay[0]) & mask;
172 s1_ptr = (s->write_ptr + s->buffer_size - s->delay[1]) & mask;
173
174 if (s->par_middle_phase)
175 mid = -mid;
176
177 side[0] = buffer[s0_ptr] * s->par_side_gain;
178 side[1] = buffer[s1_ptr] * s->par_side_gain;
179 side_l = side[0] * s->balance_l[0] - side[1] * s->balance_l[1];
180 side_r = side[1] * s->balance_r[1] - side[0] * s->balance_r[0];
181
182 dst[0] = (mid + side_l) * level_out;
183 dst[1] = (mid + side_r) * level_out;
184
185 s->write_ptr = (s->write_ptr + 1) & mask;
186 }
187
188 if (out != in)
189 av_frame_free(&in);
190 return ff_filter_frame(outlink, out);
191 }
192
193 static av_cold void uninit(AVFilterContext *ctx)
194 {
195 HaasContext *s = ctx->priv;
196
197 av_freep(&s->buffer);
198 s->buffer_size = 0;
199 }
200
201 static const AVFilterPad inputs[] = {
202 {
203 .name = "default",
204 .type = AVMEDIA_TYPE_AUDIO,
205 .filter_frame = filter_frame,
206 .config_props = config_input,
207 },
208 };
209
210 const AVFilter ff_af_haas = {
211 .name = "haas",
212 .description = NULL_IF_CONFIG_SMALL("Apply Haas Stereo Enhancer."),
213 .priv_size = sizeof(HaasContext),
214 .priv_class = &haas_class,
215 .uninit = uninit,
216 FILTER_INPUTS(inputs),
217 FILTER_OUTPUTS(ff_audio_default_filterpad),
218 FILTER_QUERY_FUNC(query_formats),
219 };
220