1. ffmpeg
  2. Commit ca5cf84655f38e8e1eaaff714d62ee824c21a309
Browse code

avfilter: add superequalizer filter

Signed-off-by: Paul B Mahol <onemda@gmail.com>

Paul B Mahol authored on 2017/06/16 21:07:01
Showing 6 changed files
Changelog
History View file @ ca5cf84
... ... 
@@ -20,6 +20,7 @@ version <next>:
20 20 
 - sofalizer filter switched to libmysofa
21 21 
 - Gremlin Digital Video demuxer and decoder
22 22 
 - headphone audio filter
23 
+- superequalizer audio filter
23 24
24 25 
 version 3.3:
25 26 
 - CrystalHD decoder moved to new decode API
doc/filters.texi
History View file @ ca5cf84
... ... 
@@ -3835,6 +3835,49 @@ channels. Default is 0.3.
3835 3835 
 Set level of input signal of original channel. Default is 0.8.
3836 3836 
 @end table
3837 3837
3838 
+@section superequalizer
3839 
+Apply 18th band equalizer.
3840 
+
3841 
+The filter accpets the following options:
3842 
+@table @option
3843 
+@item 1b
3844 
+Set 65Hz band gain.
3845 
+@item 2b
3846 
+Set 92Hz band gain.
3847 
+@item 3b
3848 
+Set 131Hz band gain.
3849 
+@item 4b
3850 
+Set 185Hz band gain.
3851 
+@item 5b
3852 
+Set 262Hz band gain.
3853 
+@item 6b
3854 
+Set 370Hz band gain.
3855 
+@item 7b
3856 
+Set 523Hz band gain.
3857 
+@item 8b
3858 
+Set 740Hz band gain.
3859 
+@item 9b
3860 
+Set 1047Hz band gain.
3861 
+@item 10b
3862 
+Set 1480Hz band gain.
3863 
+@item 11b
3864 
+Set 2093Hz band gain.
3865 
+@item 12b
3866 
+Set 2960Hz band gain.
3867 
+@item 13b
3868 
+Set 4186Hz band gain.
3869 
+@item 14b
3870 
+Set 5920Hz band gain.
3871 
+@item 15b
3872 
+Set 8372Hz band gain.
3873 
+@item 16b
3874 
+Set 11840Hz band gain.
3875 
+@item 17b
3876 
+Set 16744Hz band gain.
3877 
+@item 18b
3878 
+Set 20000Hz band gain.
3879 
+@end table
3880 
+
3838 3881 
 @section surround
3839 3882 
 Apply audio surround upmix filter.
3840 3883
libavfilter/Makefile
History View file @ ca5cf84
... ... 
@@ -109,6 +109,7 @@ OBJS-$(CONFIG_SILENCEREMOVE_FILTER)          += af_silenceremove.o
109 109 
 OBJS-$(CONFIG_SOFALIZER_FILTER)              += af_sofalizer.o
110 110 
 OBJS-$(CONFIG_STEREOTOOLS_FILTER)            += af_stereotools.o
111 111 
 OBJS-$(CONFIG_STEREOWIDEN_FILTER)            += af_stereowiden.o
112 
+OBJS-$(CONFIG_SUPEREQUALIZER_FILTER)         += af_superequalizer.o
112 113 
 OBJS-$(CONFIG_SURROUND_FILTER)               += af_surround.o
113 114 
 OBJS-$(CONFIG_TREBLE_FILTER)                 += af_biquads.o
114 115 
 OBJS-$(CONFIG_TREMOLO_FILTER)                += af_tremolo.o
libavfilter/af_superequalizer.c
History View file @ ca5cf84
115 116 
new file mode 100644
... ... 
@@ -0,0 +1,368 @@
0 
+/*
1 
+ * Copyright (c) 2002 Naoki Shibata
2 
+ * Copyright (c) 2017 Paul B Mahol
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
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+ * 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/opt.h"
22 
+
23 
+#include "libavcodec/avfft.h"
24 
+
25 
+#include "audio.h"
26 
+#include "avfilter.h"
27 
+#include "internal.h"
28 
+
29 
+#define NBANDS 17
30 
+#define M 15
31 
+
32 
+typedef struct EqParameter {
33 
+    float lower, upper, gain;
34 
+} EqParameter;
35 
+
36 
+typedef struct SuperEqualizerContext {
37 
+    const AVClass *class;
38 
+
39 
+    EqParameter params[NBANDS + 1];
40 
+
41 
+    float gains[NBANDS + 1];
42 
+
43 
+    float fact[M + 1];
44 
+    float aa;
45 
+    float iza;
46 
+    float *ires, *irest;
47 
+    float *fsamples;
48 
+    int winlen, tabsize;
49 
+
50 
+    AVFrame *in, *out;
51 
+    RDFTContext *rdft, *irdft;
52 
+} SuperEqualizerContext;
53 
+
54 
+static const float bands[] = {
55 
+    65.406392, 92.498606, 130.81278, 184.99721, 261.62557, 369.99442, 523.25113, 739.9884, 1046.5023,
56 
+    1479.9768, 2093.0045, 2959.9536, 4186.0091, 5919.9072, 8372.0181, 11839.814, 16744.036
57 
+};
58 
+
59 
+static float izero(SuperEqualizerContext *s, float x)
60 
+{
61 
+    float ret = 1;
62 
+    int m;
63 
+
64 
+    for (m = 1; m <= M; m++) {
65 
+        float t;
66 
+
67 
+        t = pow(x / 2, m) / s->fact[m];
68 
+        ret += t*t;
69 
+    }
70 
+
71 
+    return ret;
72 
+}
73 
+
74 
+static float hn_lpf(int n, float f, float fs)
75 
+{
76 
+    float t = 1 / fs;
77 
+    float omega = 2 * M_PI * f;
78 
+
79 
+    if (n * omega * t == 0)
80 
+        return 2 * f * t;
81 
+    return 2 * f * t * sinf(n * omega * t) / (n * omega * t);
82 
+}
83 
+
84 
+static float hn_imp(int n)
85 
+{
86 
+    return n == 0 ? 1.f : 0.f;
87 
+}
88 
+
89 
+static float hn(int n, EqParameter *param, float fs)
90 
+{
91 
+    float ret, lhn;
92 
+    int i;
93 
+
94 
+    lhn = hn_lpf(n, param[0].upper, fs);
95 
+    ret = param[0].gain*lhn;
96 
+
97 
+    for (i = 1; i < NBANDS + 1 && param[i].upper < fs / 2; i++) {
98 
+        float lhn2 = hn_lpf(n, param[i].upper, fs);
99 
+        ret += param[i].gain * (lhn2 - lhn);
100 
+        lhn = lhn2;
101 
+    }
102 
+
103 
+    ret += param[i].gain * (hn_imp(n) - lhn);
104 
+
105 
+    return ret;
106 
+}
107 
+
108 
+static float alpha(float a)
109 
+{
110 
+    if (a <= 21)
111 
+        return 0;
112 
+    if (a <= 50)
113 
+        return .5842f * pow(a - 21, 0.4f) + 0.07886f * (a - 21);
114 
+    return .1102f * (a - 8.7f);
115 
+}
116 
+
117 
+static float win(SuperEqualizerContext *s, float n, int N)
118 
+{
119 
+    return izero(s, alpha(s->aa) * sqrtf(1 - 4 * n * n / ((N - 1) * (N - 1)))) / s->iza;
120 
+}
121 
+
122 
+static void process_param(float *bc, EqParameter *param, float fs)
123 
+{
124 
+    int i;
125 
+
126 
+    for (i = 0; i <= NBANDS; i++) {
127 
+        param[i].lower = i == 0 ? 0 : bands[i - 1];
128 
+        param[i].upper = i == NBANDS - 1 ? fs : bands[i];
129 
+        param[i].gain  = bc[i];
130 
+    }
131 
+}
132 
+
133 
+static int equ_init(SuperEqualizerContext *s, int wb)
134 
+{
135 
+    int i,j;
136 
+
137 
+    s->rdft  = av_rdft_init(wb, DFT_R2C);
138 
+    s->irdft = av_rdft_init(wb, IDFT_C2R);
139 
+    if (!s->rdft || !s->irdft)
140 
+        return AVERROR(ENOMEM);
141 
+
142 
+    s->aa = 96;
143 
+    s->winlen = (1 << (wb-1))-1;
144 
+    s->tabsize  = 1 << wb;
145 
+
146 
+    s->ires     = av_calloc(s->tabsize, sizeof(float));
147 
+    s->irest    = av_calloc(s->tabsize, sizeof(float));
148 
+    s->fsamples = av_calloc(s->tabsize, sizeof(float));
149 
+
150 
+    for (i = 0; i <= M; i++) {
151 
+        s->fact[i] = 1;
152 
+        for (j = 1; j <= i; j++)
153 
+            s->fact[i] *= j;
154 
+    }
155 
+
156 
+    s->iza = izero(s, alpha(s->aa));
157 
+
158 
+    return 0;
159 
+}
160 
+
161 
+static void make_fir(SuperEqualizerContext *s, float *lbc, float *rbc, EqParameter *param, float fs)
162 
+{
163 
+    const int winlen = s->winlen;
164 
+    const int tabsize = s->tabsize;
165 
+    float *nires;
166 
+    int i;
167 
+
168 
+    if (fs <= 0)
169 
+        return;
170 
+
171 
+    process_param(lbc, param, fs);
172 
+    for (i = 0; i < winlen; i++)
173 
+        s->irest[i] = hn(i - winlen / 2, param, fs) * win(s, i - winlen / 2, winlen);
174 
+    for (; i < tabsize; i++)
175 
+        s->irest[i] = 0;
176 
+
177 
+    av_rdft_calc(s->rdft, s->irest);
178 
+    nires = s->ires;
179 
+    for (i = 0; i < tabsize; i++)
180 
+        nires[i] = s->irest[i];
181 
+}
182 
+
183 
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
184 
+{
185 
+    AVFilterContext *ctx = inlink->dst;
186 
+    SuperEqualizerContext *s = ctx->priv;
187 
+    AVFilterLink *outlink = ctx->outputs[0];
188 
+    const float *ires = s->ires;
189 
+    float *fsamples = s->fsamples;
190 
+    int ch, i;
191 
+
192 
+    AVFrame *out = ff_get_audio_buffer(outlink, s->winlen);
193 
+    float *src, *dst, *ptr;
194 
+
195 
+    if (!out) {
196 
+        av_frame_free(&in);
197 
+        return AVERROR(ENOMEM);
198 
+    }
199 
+
200 
+    for (ch = 0; ch < in->channels; ch++) {
201 
+        ptr = (float *)out->extended_data[ch];
202 
+        dst = (float *)s->out->extended_data[ch];
203 
+        src = (float *)in->extended_data[ch];
204 
+
205 
+        for (i = 0; i < s->winlen; i++)
206 
+            fsamples[i] = src[i];
207 
+        for (; i < s->tabsize; i++)
208 
+            fsamples[i] = 0;
209 
+
210 
+        av_rdft_calc(s->rdft, fsamples);
211 
+
212 
+        fsamples[0] = ires[0] * fsamples[0];
213 
+        fsamples[1] = ires[1] * fsamples[1];
214 
+        for (i = 1; i < s->tabsize / 2; i++) {
215 
+            float re, im;
216 
+
217 
+            re = ires[i*2  ] * fsamples[i*2] - ires[i*2+1] * fsamples[i*2+1];
218 
+            im = ires[i*2+1] * fsamples[i*2] + ires[i*2  ] * fsamples[i*2+1];
219 
+
220 
+            fsamples[i*2  ] = re;
221 
+            fsamples[i*2+1] = im;
222 
+        }
223 
+
224 
+        av_rdft_calc(s->irdft, fsamples);
225 
+
226 
+        for (i = 0; i < s->winlen; i++)
227 
+            dst[i] += fsamples[i] / s->tabsize * 2;
228 
+        for (i = s->winlen; i < s->tabsize; i++)
229 
+            dst[i]  = fsamples[i] / s->tabsize * 2;
230 
+        for (i = 0; i < s->winlen; i++)
231 
+            ptr[i] = dst[i];
232 
+        for (i = 0; i < s->winlen; i++)
233 
+            dst[i] = dst[i+s->winlen];
234 
+    }
235 
+
236 
+    out->pts = in->pts;
237 
+    av_frame_free(&in);
238 
+
239 
+    return ff_filter_frame(outlink, out);
240 
+}
241 
+
242 
+static av_cold int init(AVFilterContext *ctx)
243 
+{
244 
+    SuperEqualizerContext *s = ctx->priv;
245 
+
246 
+    return equ_init(s, 14);
247 
+}
248 
+
249 
+static int query_formats(AVFilterContext *ctx)
250 
+{
251 
+    AVFilterFormats *formats;
252 
+    AVFilterChannelLayouts *layouts;
253 
+    static const enum AVSampleFormat sample_fmts[] = {
254 
+        AV_SAMPLE_FMT_FLTP,
255 
+        AV_SAMPLE_FMT_NONE
256 
+    };
257 
+    int ret;
258 
+
259 
+    layouts = ff_all_channel_counts();
260 
+    if (!layouts)
261 
+        return AVERROR(ENOMEM);
262 
+    ret = ff_set_common_channel_layouts(ctx, layouts);
263 
+    if (ret < 0)
264 
+        return ret;
265 
+
266 
+    formats = ff_make_format_list(sample_fmts);
267 
+    if ((ret = ff_set_common_formats(ctx, formats)) < 0)
268 
+        return ret;
269 
+
270 
+    formats = ff_all_samplerates();
271 
+    return ff_set_common_samplerates(ctx, formats);
272 
+}
273 
+
274 
+static int config_input(AVFilterLink *inlink)
275 
+{
276 
+    AVFilterContext *ctx = inlink->dst;
277 
+    SuperEqualizerContext *s = ctx->priv;
278 
+
279 
+    inlink->partial_buf_size =
280 
+    inlink->min_samples =
281 
+    inlink->max_samples = s->winlen;
282 
+
283 
+    s->out = ff_get_audio_buffer(inlink, s->tabsize);
284 
+    if (!s->out)
285 
+        return AVERROR(ENOMEM);
286 
+
287 
+    return 0;
288 
+}
289 
+
290 
+static int config_output(AVFilterLink *outlink)
291 
+{
292 
+    AVFilterContext *ctx = outlink->src;
293 
+    SuperEqualizerContext *s = ctx->priv;
294 
+
295 
+    make_fir(s, s->gains, s->gains, s->params, outlink->sample_rate);
296 
+
297 
+    return 0;
298 
+}
299 
+
300 
+static av_cold void uninit(AVFilterContext *ctx)
301 
+{
302 
+    SuperEqualizerContext *s = ctx->priv;
303 
+
304 
+    av_freep(&s->irest);
305 
+    av_freep(&s->ires);
306 
+    av_freep(&s->fsamples);
307 
+    av_rdft_end(s->rdft);
308 
+    av_rdft_end(s->irdft);
309 
+}
310 
+
311 
+static const AVFilterPad superequalizer_inputs[] = {
312 
+    {
313 
+        .name         = "default",
314 
+        .type         = AVMEDIA_TYPE_AUDIO,
315 
+        .filter_frame = filter_frame,
316 
+        .config_props = config_input,
317 
+    },
318 
+    { NULL }
319 
+};
320 
+
321 
+static const AVFilterPad superequalizer_outputs[] = {
322 
+    {
323 
+        .name         = "default",
324 
+        .type         = AVMEDIA_TYPE_AUDIO,
325 
+        .config_props = config_output,
326 
+    },
327 
+    { NULL }
328 
+};
329 
+
330 
+#define AF AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
331 
+#define OFFSET(x) offsetof(SuperEqualizerContext, x)
332 
+
333 
+static const AVOption superequalizer_options[] = {
334 
+    {  "1b", "set 65Hz band gain",    OFFSET(gains [0]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
335 
+    {  "2b", "set 92Hz band gain",    OFFSET(gains [1]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
336 
+    {  "3b", "set 131Hz band gain",   OFFSET(gains [2]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
337 
+    {  "4b", "set 185Hz band gain",   OFFSET(gains [3]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
338 
+    {  "5b", "set 262Hz band gain",   OFFSET(gains [4]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
339 
+    {  "6b", "set 370Hz band gain",   OFFSET(gains [5]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
340 
+    {  "7b", "set 523Hz band gain",   OFFSET(gains [6]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
341 
+    {  "8b", "set 740Hz band gain",   OFFSET(gains [7]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
342 
+    {  "9b", "set 1047Hz band gain",  OFFSET(gains [8]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
343 
+    { "10b", "set 1480Hz band gain",  OFFSET(gains [9]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
344 
+    { "11b", "set 2093Hz band gain",  OFFSET(gains[10]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
345 
+    { "12b", "set 2960Hz band gain",  OFFSET(gains[11]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
346 
+    { "13b", "set 4186Hz band gain",  OFFSET(gains[12]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
347 
+    { "14b", "set 5920Hz band gain",  OFFSET(gains[13]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
348 
+    { "15b", "set 8372Hz band gain",  OFFSET(gains[14]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
349 
+    { "16b", "set 11840Hz band gain", OFFSET(gains[15]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
350 
+    { "17b", "set 16744Hz band gain", OFFSET(gains[16]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
351 
+    { "18b", "set 20000Hz band gain", OFFSET(gains[17]), AV_OPT_TYPE_FLOAT, {.dbl=1}, 0, 20, AF },
352 
+    { NULL }
353 
+};
354 
+
355 
+AVFILTER_DEFINE_CLASS(superequalizer);
356 
+
357 
+AVFilter ff_af_superequalizer = {
358 
+    .name          = "superequalizer",
359 
+    .description   = NULL_IF_CONFIG_SMALL("Apply 18-th band equalization filter."),
360 
+    .priv_size     = sizeof(SuperEqualizerContext),
361 
+    .priv_class    = &superequalizer_class,
362 
+    .query_formats = query_formats,
363 
+    .init          = init,
364 
+    .uninit        = uninit,
365 
+    .inputs        = superequalizer_inputs,
366 
+    .outputs       = superequalizer_outputs,
367 
+};
libavfilter/allfilters.c
History View file @ ca5cf84
... ... 
@@ -122,6 +122,7 @@ static void register_all(void)
122 122 
     REGISTER_FILTER(SOFALIZER,      sofalizer,      af);
123 123 
     REGISTER_FILTER(STEREOTOOLS,    stereotools,    af);
124 124 
     REGISTER_FILTER(STEREOWIDEN,    stereowiden,    af);
125 
+    REGISTER_FILTER(SUPEREQUALIZER, superequalizer, af);
125 126 
     REGISTER_FILTER(SURROUND,       surround,       af);
126 127 
     REGISTER_FILTER(TREBLE,         treble,         af);
127 128 
     REGISTER_FILTER(TREMOLO,        tremolo,        af);
libavfilter/version.h
History View file @ ca5cf84
... ... 
@@ -30,7 +30,7 @@
30 30 
 #include "libavutil/version.h"
31 31
32 32 
 #define LIBAVFILTER_VERSION_MAJOR   6
33 
-#define LIBAVFILTER_VERSION_MINOR  92
33 
+#define LIBAVFILTER_VERSION_MINOR  93
34 34 
 #define LIBAVFILTER_VERSION_MICRO 100
35 35
36 36 
 #define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \