1 /*
2 * This file is part of FFmpeg.
3 *
4 * FFmpeg is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
8 *
9 * FFmpeg is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Lesser General Public License for more details.
13 *
14 * You should have received a copy of the GNU Lesser General Public
15 * License along with FFmpeg; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
19 #include "libavutil/channel_layout.h"
20 #include "libavutil/ffmath.h"
21 #include "libavutil/opt.h"
22 #include "avfilter.h"
23 #include "audio.h"
24 #include "formats.h"
25
26 typedef struct ASubBoostContext {
27 const AVClass *class;
28
29 double dry_gain;
30 double wet_gain;
31 double feedback;
32 double decay;
33 double delay;
34 double cutoff;
35 double slope;
36
37 double a0, a1, a2;
38 double b0, b1, b2;
39
40 int *write_pos;
41 int buffer_samples;
42
43 AVFrame *w;
44 AVFrame *buffer;
45 } ASubBoostContext;
46
query_formats(AVFilterContext * ctx)47 static int query_formats(AVFilterContext *ctx)
48 {
49 AVFilterFormats *formats = NULL;
50 AVFilterChannelLayouts *layouts = NULL;
51 static const enum AVSampleFormat sample_fmts[] = {
52 AV_SAMPLE_FMT_DBLP,
53 AV_SAMPLE_FMT_NONE
54 };
55 int ret;
56
57 formats = ff_make_format_list(sample_fmts);
58 if (!formats)
59 return AVERROR(ENOMEM);
60 ret = ff_set_common_formats(ctx, formats);
61 if (ret < 0)
62 return ret;
63
64 layouts = ff_all_channel_counts();
65 if (!layouts)
66 return AVERROR(ENOMEM);
67
68 ret = ff_set_common_channel_layouts(ctx, layouts);
69 if (ret < 0)
70 return ret;
71
72 formats = ff_all_samplerates();
73 return ff_set_common_samplerates(ctx, formats);
74 }
75
get_coeffs(AVFilterContext * ctx)76 static int get_coeffs(AVFilterContext *ctx)
77 {
78 ASubBoostContext *s = ctx->priv;
79 AVFilterLink *inlink = ctx->inputs[0];
80 double w0 = 2 * M_PI * s->cutoff / inlink->sample_rate;
81 double alpha = sin(w0) / 2 * sqrt(2. * (1. / s->slope - 1.) + 2.);
82
83 s->a0 = 1 + alpha;
84 s->a1 = -2 * cos(w0);
85 s->a2 = 1 - alpha;
86 s->b0 = (1 - cos(w0)) / 2;
87 s->b1 = 1 - cos(w0);
88 s->b2 = (1 - cos(w0)) / 2;
89
90 s->a1 /= s->a0;
91 s->a2 /= s->a0;
92 s->b0 /= s->a0;
93 s->b1 /= s->a0;
94 s->b2 /= s->a0;
95
96 s->buffer_samples = inlink->sample_rate * s->delay / 1000;
97
98 return 0;
99 }
100
config_input(AVFilterLink * inlink)101 static int config_input(AVFilterLink *inlink)
102 {
103 AVFilterContext *ctx = inlink->dst;
104 ASubBoostContext *s = ctx->priv;
105
106 s->buffer = ff_get_audio_buffer(inlink, inlink->sample_rate / 10);
107 s->w = ff_get_audio_buffer(inlink, 2);
108 s->write_pos = av_calloc(inlink->channels, sizeof(*s->write_pos));
109 if (!s->buffer || !s->w || !s->write_pos)
110 return AVERROR(ENOMEM);
111
112 return get_coeffs(ctx);
113 }
114
115 typedef struct ThreadData {
116 AVFrame *in, *out;
117 } ThreadData;
118
filter_channels(AVFilterContext * ctx,void * arg,int jobnr,int nb_jobs)119 static int filter_channels(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
120 {
121 ASubBoostContext *s = ctx->priv;
122 ThreadData *td = arg;
123 AVFrame *out = td->out;
124 AVFrame *in = td->in;
125 const double mix = ctx->is_disabled ? 0. : 1.;
126 const double wet = ctx->is_disabled ? 1. : s->wet_gain;
127 const double dry = ctx->is_disabled ? 1. : s->dry_gain;
128 const double feedback = s->feedback, decay = s->decay;
129 const double b0 = s->b0;
130 const double b1 = s->b1;
131 const double b2 = s->b2;
132 const double a1 = -s->a1;
133 const double a2 = -s->a2;
134 const int start = (in->channels * jobnr) / nb_jobs;
135 const int end = (in->channels * (jobnr+1)) / nb_jobs;
136 const int buffer_samples = s->buffer_samples;
137
138 for (int ch = start; ch < end; ch++) {
139 const double *src = (const double *)in->extended_data[ch];
140 double *dst = (double *)out->extended_data[ch];
141 double *buffer = (double *)s->buffer->extended_data[ch];
142 double *w = (double *)s->w->extended_data[ch];
143 int write_pos = s->write_pos[ch];
144
145 for (int n = 0; n < in->nb_samples; n++) {
146 double out_sample;
147
148 out_sample = src[n] * b0 + w[0];
149 w[0] = b1 * src[n] + w[1] + a1 * out_sample;
150 w[1] = b2 * src[n] + a2 * out_sample;
151
152 buffer[write_pos] = buffer[write_pos] * decay + out_sample * feedback;
153 dst[n] = (src[n] * dry + buffer[write_pos] * mix) * wet;
154
155 if (++write_pos >= buffer_samples)
156 write_pos = 0;
157 }
158
159 s->write_pos[ch] = write_pos;
160 }
161
162 return 0;
163 }
164
filter_frame(AVFilterLink * inlink,AVFrame * in)165 static int filter_frame(AVFilterLink *inlink, AVFrame *in)
166 {
167 AVFilterContext *ctx = inlink->dst;
168 AVFilterLink *outlink = ctx->outputs[0];
169 ThreadData td;
170 AVFrame *out;
171
172 if (av_frame_is_writable(in)) {
173 out = in;
174 } else {
175 out = ff_get_audio_buffer(outlink, in->nb_samples);
176 if (!out) {
177 av_frame_free(&in);
178 return AVERROR(ENOMEM);
179 }
180 av_frame_copy_props(out, in);
181 }
182
183 td.in = in; td.out = out;
184 ctx->internal->execute(ctx, filter_channels, &td, NULL, FFMIN(inlink->channels,
185 ff_filter_get_nb_threads(ctx)));
186
187 if (out != in)
188 av_frame_free(&in);
189 return ff_filter_frame(outlink, out);
190 }
191
uninit(AVFilterContext * ctx)192 static av_cold void uninit(AVFilterContext *ctx)
193 {
194 ASubBoostContext *s = ctx->priv;
195
196 av_frame_free(&s->buffer);
197 av_frame_free(&s->w);
198 av_freep(&s->write_pos);
199 }
200
process_command(AVFilterContext * ctx,const char * cmd,const char * args,char * res,int res_len,int flags)201 static int process_command(AVFilterContext *ctx, const char *cmd, const char *args,
202 char *res, int res_len, int flags)
203 {
204 int ret;
205
206 ret = ff_filter_process_command(ctx, cmd, args, res, res_len, flags);
207 if (ret < 0)
208 return ret;
209
210 return get_coeffs(ctx);
211 }
212
213 #define OFFSET(x) offsetof(ASubBoostContext, x)
214 #define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_RUNTIME_PARAM
215
216 static const AVOption asubboost_options[] = {
217 { "dry", "set dry gain", OFFSET(dry_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
218 { "wet", "set wet gain", OFFSET(wet_gain), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
219 { "decay", "set decay", OFFSET(decay), AV_OPT_TYPE_DOUBLE, {.dbl=0.7}, 0, 1, FLAGS },
220 { "feedback", "set feedback", OFFSET(feedback), AV_OPT_TYPE_DOUBLE, {.dbl=0.9}, 0, 1, FLAGS },
221 { "cutoff", "set cutoff", OFFSET(cutoff), AV_OPT_TYPE_DOUBLE, {.dbl=100}, 50, 900, FLAGS },
222 { "slope", "set slope", OFFSET(slope), AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0.0001, 1, FLAGS },
223 { "delay", "set delay", OFFSET(delay), AV_OPT_TYPE_DOUBLE, {.dbl=20}, 1, 100, FLAGS },
224 { NULL }
225 };
226
227 AVFILTER_DEFINE_CLASS(asubboost);
228
229 static const AVFilterPad inputs[] = {
230 {
231 .name = "default",
232 .type = AVMEDIA_TYPE_AUDIO,
233 .filter_frame = filter_frame,
234 .config_props = config_input,
235 },
236 { NULL }
237 };
238
239 static const AVFilterPad outputs[] = {
240 {
241 .name = "default",
242 .type = AVMEDIA_TYPE_AUDIO,
243 },
244 { NULL }
245 };
246
247 AVFilter ff_af_asubboost = {
248 .name = "asubboost",
249 .description = NULL_IF_CONFIG_SMALL("Boost subwoofer frequencies."),
250 .query_formats = query_formats,
251 .priv_size = sizeof(ASubBoostContext),
252 .priv_class = &asubboost_class,
253 .uninit = uninit,
254 .inputs = inputs,
255 .outputs = outputs,
256 .process_command = process_command,
257 .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL |
258 AVFILTER_FLAG_SLICE_THREADS,
259 };
260