1 /*
2 * GStreamer
3 * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
4 *
5 * This library is free software; you can redistribute it and/or
6 * modify it under the terms of the GNU Library General Public
7 * License as published by the Free Software Foundation; either
8 * version 2 of the License, or (at your option) any later version.
9 *
10 * This library is distributed in the hope that it will be useful,
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
13 * Library General Public License for more details.
14 *
15 * You should have received a copy of the GNU Library General Public
16 * License along with this library; if not, write to the
17 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
18 * Boston, MA 02110-1301, USA.
19 */
20
21 /**
22 * SECTION:element-audiodynamic
23 * @title: audiodynamic
24 *
25 * This element can act as a compressor or expander. A compressor changes the
26 * amplitude of all samples above a specific threshold with a specific ratio,
27 * a expander does the same for all samples below a specific threshold. If
28 * soft-knee mode is selected the ratio is applied smoothly.
29 *
30 * ## Example launch line
31 * |[
32 * gst-launch-1.0 audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 ratio=0.5 ! alsasink
33 * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 ratio=4.0 ! alsasink
34 * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
35 * ]|
36 *
37 */
38
39 /* TODO: Implement attack and release parameters */
40
41 #ifdef HAVE_CONFIG_H
42 #include "config.h"
43 #endif
44
45 #include <gst/gst.h>
46 #include <gst/base/gstbasetransform.h>
47 #include <gst/audio/audio.h>
48 #include <gst/audio/gstaudiofilter.h>
49
50 #include "audiodynamic.h"
51
52 #define GST_CAT_DEFAULT gst_audio_dynamic_debug
53 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
54
55 /* Filter signals and args */
56 enum
57 {
58 /* FILL ME */
59 LAST_SIGNAL
60 };
61
62 enum
63 {
64 PROP_0,
65 PROP_CHARACTERISTICS,
66 PROP_MODE,
67 PROP_THRESHOLD,
68 PROP_RATIO
69 };
70
71 #define ALLOWED_CAPS \
72 "audio/x-raw," \
73 " format=(string) {"GST_AUDIO_NE(S16)","GST_AUDIO_NE(F32)"}," \
74 " rate=(int)[1,MAX]," \
75 " channels=(int)[1,MAX]," \
76 " layout=(string) {interleaved, non-interleaved}"
77
78 G_DEFINE_TYPE (GstAudioDynamic, gst_audio_dynamic, GST_TYPE_AUDIO_FILTER);
79 GST_ELEMENT_REGISTER_DEFINE (audiodynamic, "audiodynamic",
80 GST_RANK_NONE, GST_TYPE_AUDIO_DYNAMIC);
81
82 static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,
83 const GValue * value, GParamSpec * pspec);
84 static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,
85 GValue * value, GParamSpec * pspec);
86
87 static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,
88 const GstAudioInfo * info);
89 static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,
90 GstBuffer * buf);
91
92 static void
93 gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
94 gint16 * data, guint num_samples);
95 static void
96 gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
97 filter, gfloat * data, guint num_samples);
98 static void
99 gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
100 gint16 * data, guint num_samples);
101 static void
102 gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
103 filter, gfloat * data, guint num_samples);
104 static void gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic
105 * filter, gint16 * data, guint num_samples);
106 static void
107 gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
108 gfloat * data, guint num_samples);
109 static void gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic
110 * filter, gint16 * data, guint num_samples);
111 static void
112 gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
113 gfloat * data, guint num_samples);
114
115 static const GstAudioDynamicProcessFunc process_functions[] = {
116 (GstAudioDynamicProcessFunc)
117 gst_audio_dynamic_transform_hard_knee_compressor_int,
118 (GstAudioDynamicProcessFunc)
119 gst_audio_dynamic_transform_hard_knee_compressor_float,
120 (GstAudioDynamicProcessFunc)
121 gst_audio_dynamic_transform_soft_knee_compressor_int,
122 (GstAudioDynamicProcessFunc)
123 gst_audio_dynamic_transform_soft_knee_compressor_float,
124 (GstAudioDynamicProcessFunc)
125 gst_audio_dynamic_transform_hard_knee_expander_int,
126 (GstAudioDynamicProcessFunc)
127 gst_audio_dynamic_transform_hard_knee_expander_float,
128 (GstAudioDynamicProcessFunc)
129 gst_audio_dynamic_transform_soft_knee_expander_int,
130 (GstAudioDynamicProcessFunc)
131 gst_audio_dynamic_transform_soft_knee_expander_float
132 };
133
134 enum
135 {
136 CHARACTERISTICS_HARD_KNEE = 0,
137 CHARACTERISTICS_SOFT_KNEE
138 };
139
140 #define GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS (gst_audio_dynamic_characteristics_get_type ())
141 static GType
gst_audio_dynamic_characteristics_get_type(void)142 gst_audio_dynamic_characteristics_get_type (void)
143 {
144 static GType gtype = 0;
145
146 if (gtype == 0) {
147 static const GEnumValue values[] = {
148 {CHARACTERISTICS_HARD_KNEE, "Hard Knee (default)",
149 "hard-knee"},
150 {CHARACTERISTICS_SOFT_KNEE, "Soft Knee (smooth)",
151 "soft-knee"},
152 {0, NULL, NULL}
153 };
154
155 gtype = g_enum_register_static ("GstAudioDynamicCharacteristics", values);
156 }
157 return gtype;
158 }
159
160 enum
161 {
162 MODE_COMPRESSOR = 0,
163 MODE_EXPANDER
164 };
165
166 #define GST_TYPE_AUDIO_DYNAMIC_MODE (gst_audio_dynamic_mode_get_type ())
167 static GType
gst_audio_dynamic_mode_get_type(void)168 gst_audio_dynamic_mode_get_type (void)
169 {
170 static GType gtype = 0;
171
172 if (gtype == 0) {
173 static const GEnumValue values[] = {
174 {MODE_COMPRESSOR, "Compressor (default)",
175 "compressor"},
176 {MODE_EXPANDER, "Expander", "expander"},
177 {0, NULL, NULL}
178 };
179
180 gtype = g_enum_register_static ("GstAudioDynamicMode", values);
181 }
182 return gtype;
183 }
184
185 static void
gst_audio_dynamic_set_process_function(GstAudioDynamic * filter,const GstAudioInfo * info)186 gst_audio_dynamic_set_process_function (GstAudioDynamic * filter,
187 const GstAudioInfo * info)
188 {
189 gint func_index;
190
191 func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;
192 func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;
193 func_index += (GST_AUDIO_INFO_FORMAT (info) == GST_AUDIO_FORMAT_F32) ? 1 : 0;
194
195 g_assert (func_index >= 0 && func_index < G_N_ELEMENTS (process_functions));
196
197 filter->process = process_functions[func_index];
198 }
199
200 /* GObject vmethod implementations */
201
202 static void
gst_audio_dynamic_class_init(GstAudioDynamicClass * klass)203 gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
204 {
205 GObjectClass *gobject_class;
206 GstElementClass *gstelement_class;
207 GstCaps *caps;
208
209 GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0,
210 "audiodynamic element");
211
212 gobject_class = (GObjectClass *) klass;
213 gstelement_class = (GstElementClass *) klass;
214
215 gobject_class->set_property = gst_audio_dynamic_set_property;
216 gobject_class->get_property = gst_audio_dynamic_get_property;
217
218 g_object_class_install_property (gobject_class, PROP_CHARACTERISTICS,
219 g_param_spec_enum ("characteristics", "Characteristics",
220 "Selects whether the ratio should be applied smooth (soft-knee) "
221 "or hard (hard-knee).",
222 GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, CHARACTERISTICS_HARD_KNEE,
223 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
224
225 g_object_class_install_property (gobject_class, PROP_MODE,
226 g_param_spec_enum ("mode", "Mode",
227 "Selects whether the filter should work on loud samples (compressor) or"
228 "quiet samples (expander).",
229 GST_TYPE_AUDIO_DYNAMIC_MODE, MODE_COMPRESSOR,
230 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
231
232 g_object_class_install_property (gobject_class, PROP_THRESHOLD,
233 g_param_spec_float ("threshold", "Threshold",
234 "Threshold until the filter is activated", 0.0, 1.0,
235 0.0,
236 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
237
238 g_object_class_install_property (gobject_class, PROP_RATIO,
239 g_param_spec_float ("ratio", "Ratio",
240 "Ratio that should be applied", 0.0, G_MAXFLOAT,
241 1.0,
242 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
243
244 gst_element_class_set_static_metadata (gstelement_class,
245 "Dynamic range controller", "Filter/Effect/Audio",
246 "Compressor and Expander", "Sebastian Dröge <slomo@circular-chaos.org>");
247
248 caps = gst_caps_from_string (ALLOWED_CAPS);
249 gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
250 caps);
251 gst_caps_unref (caps);
252
253 GST_AUDIO_FILTER_CLASS (klass)->setup =
254 GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);
255
256 GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
257 GST_DEBUG_FUNCPTR (gst_audio_dynamic_transform_ip);
258 GST_BASE_TRANSFORM_CLASS (klass)->transform_ip_on_passthrough = FALSE;
259
260 gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, 0);
261 gst_type_mark_as_plugin_api (GST_TYPE_AUDIO_DYNAMIC_MODE, 0);
262 }
263
264 static void
gst_audio_dynamic_init(GstAudioDynamic * filter)265 gst_audio_dynamic_init (GstAudioDynamic * filter)
266 {
267 filter->ratio = 1.0;
268 filter->threshold = 0.0;
269 filter->characteristics = CHARACTERISTICS_HARD_KNEE;
270 filter->mode = MODE_COMPRESSOR;
271 gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
272 gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
273 }
274
275 static void
gst_audio_dynamic_set_property(GObject * object,guint prop_id,const GValue * value,GParamSpec * pspec)276 gst_audio_dynamic_set_property (GObject * object, guint prop_id,
277 const GValue * value, GParamSpec * pspec)
278 {
279 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
280
281 switch (prop_id) {
282 case PROP_CHARACTERISTICS:
283 filter->characteristics = g_value_get_enum (value);
284 gst_audio_dynamic_set_process_function (filter,
285 GST_AUDIO_FILTER_INFO (filter));
286 break;
287 case PROP_MODE:
288 filter->mode = g_value_get_enum (value);
289 gst_audio_dynamic_set_process_function (filter,
290 GST_AUDIO_FILTER_INFO (filter));
291 break;
292 case PROP_THRESHOLD:
293 filter->threshold = g_value_get_float (value);
294 break;
295 case PROP_RATIO:
296 filter->ratio = g_value_get_float (value);
297 break;
298 default:
299 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
300 break;
301 }
302 }
303
304 static void
gst_audio_dynamic_get_property(GObject * object,guint prop_id,GValue * value,GParamSpec * pspec)305 gst_audio_dynamic_get_property (GObject * object, guint prop_id,
306 GValue * value, GParamSpec * pspec)
307 {
308 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
309
310 switch (prop_id) {
311 case PROP_CHARACTERISTICS:
312 g_value_set_enum (value, filter->characteristics);
313 break;
314 case PROP_MODE:
315 g_value_set_enum (value, filter->mode);
316 break;
317 case PROP_THRESHOLD:
318 g_value_set_float (value, filter->threshold);
319 break;
320 case PROP_RATIO:
321 g_value_set_float (value, filter->ratio);
322 break;
323 default:
324 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
325 break;
326 }
327 }
328
329 /* GstAudioFilter vmethod implementations */
330
331 static gboolean
gst_audio_dynamic_setup(GstAudioFilter * base,const GstAudioInfo * info)332 gst_audio_dynamic_setup (GstAudioFilter * base, const GstAudioInfo * info)
333 {
334 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
335
336 gst_audio_dynamic_set_process_function (filter, info);
337 return TRUE;
338 }
339
340 static void
gst_audio_dynamic_transform_hard_knee_compressor_int(GstAudioDynamic * filter,gint16 * data,guint num_samples)341 gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
342 gint16 * data, guint num_samples)
343 {
344 glong val;
345 glong thr_p = filter->threshold * G_MAXINT16;
346 glong thr_n = filter->threshold * G_MININT16;
347
348 /* Nothing to do for us if ratio is 1.0 or if the threshold
349 * equals 1.0. */
350 if (filter->threshold == 1.0 || filter->ratio == 1.0)
351 return;
352
353 for (; num_samples; num_samples--) {
354 val = *data;
355
356 if (val > thr_p) {
357 val = thr_p + (val - thr_p) * filter->ratio;
358 } else if (val < thr_n) {
359 val = thr_n + (val - thr_n) * filter->ratio;
360 }
361 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
362 }
363 }
364
365 static void
gst_audio_dynamic_transform_hard_knee_compressor_float(GstAudioDynamic * filter,gfloat * data,guint num_samples)366 gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
367 filter, gfloat * data, guint num_samples)
368 {
369 gdouble val, threshold = filter->threshold;
370
371 /* Nothing to do for us if ratio == 1.0.
372 * As float values can be above 1.0 we have to do something
373 * if threshold is greater than 1.0. */
374 if (filter->ratio == 1.0)
375 return;
376
377 for (; num_samples; num_samples--) {
378 val = *data;
379
380 if (val > threshold) {
381 val = threshold + (val - threshold) * filter->ratio;
382 } else if (val < -threshold) {
383 val = -threshold + (val + threshold) * filter->ratio;
384 }
385 *data++ = (gfloat) val;
386 }
387 }
388
389 static void
gst_audio_dynamic_transform_soft_knee_compressor_int(GstAudioDynamic * filter,gint16 * data,guint num_samples)390 gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
391 gint16 * data, guint num_samples)
392 {
393 glong val;
394 glong thr_p = filter->threshold * G_MAXINT16;
395 glong thr_n = filter->threshold * G_MININT16;
396 gdouble a_p, b_p, c_p;
397 gdouble a_n, b_n, c_n;
398
399 /* Nothing to do for us if ratio is 1.0 or if the threshold
400 * equals 1.0. */
401 if (filter->threshold == 1.0 || filter->ratio == 1.0)
402 return;
403
404 /* We build a 2nd degree polynomial here for
405 * values greater than threshold or small than
406 * -threshold with:
407 * f(t) = t, f'(t) = 1, f'(m) = r
408 * =>
409 * a = (1-r)/(2*(t-m))
410 * b = (r*t - m)/(t-m)
411 * c = t * (1 - b - a*t)
412 * f(x) = ax^2 + bx + c
413 */
414
415 /* shouldn't happen because this would only be the case
416 * for threshold == 1.0 which we catch above */
417 g_assert (thr_p - G_MAXINT16 != 0);
418 g_assert (thr_n - G_MININT != 0);
419
420 a_p = (1 - filter->ratio) / (2 * (thr_p - G_MAXINT16));
421 b_p = (filter->ratio * thr_p - G_MAXINT16) / (thr_p - G_MAXINT16);
422 c_p = thr_p * (1 - b_p - a_p * thr_p);
423 a_n = (1 - filter->ratio) / (2 * (thr_n - G_MININT16));
424 b_n = (filter->ratio * thr_n - G_MININT16) / (thr_n - G_MININT16);
425 c_n = thr_n * (1 - b_n - a_n * thr_n);
426
427 for (; num_samples; num_samples--) {
428 val = *data;
429
430 if (val > thr_p) {
431 val = a_p * val * val + b_p * val + c_p;
432 } else if (val < thr_n) {
433 val = a_n * val * val + b_n * val + c_n;
434 }
435 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
436 }
437 }
438
439 static void
gst_audio_dynamic_transform_soft_knee_compressor_float(GstAudioDynamic * filter,gfloat * data,guint num_samples)440 gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
441 filter, gfloat * data, guint num_samples)
442 {
443 gdouble val;
444 gdouble threshold = filter->threshold;
445 gdouble a_p, b_p, c_p;
446 gdouble a_n, b_n, c_n;
447
448 /* Nothing to do for us if ratio == 1.0.
449 * As float values can be above 1.0 we have to do something
450 * if threshold is greater than 1.0. */
451 if (filter->ratio == 1.0)
452 return;
453
454 /* We build a 2nd degree polynomial here for
455 * values greater than threshold or small than
456 * -threshold with:
457 * f(t) = t, f'(t) = 1, f'(m) = r
458 * =>
459 * a = (1-r)/(2*(t-m))
460 * b = (r*t - m)/(t-m)
461 * c = t * (1 - b - a*t)
462 * f(x) = ax^2 + bx + c
463 */
464
465 /* FIXME: If threshold is the same as the maximum
466 * we need to raise it a bit to prevent
467 * division by zero. */
468 if (threshold == 1.0)
469 threshold = 1.0 + 0.00001;
470
471 a_p = (1.0 - filter->ratio) / (2.0 * (threshold - 1.0));
472 b_p = (filter->ratio * threshold - 1.0) / (threshold - 1.0);
473 c_p = threshold * (1.0 - b_p - a_p * threshold);
474 a_n = (1.0 - filter->ratio) / (2.0 * (-threshold + 1.0));
475 b_n = (-filter->ratio * threshold + 1.0) / (-threshold + 1.0);
476 c_n = -threshold * (1.0 - b_n + a_n * threshold);
477
478 for (; num_samples; num_samples--) {
479 val = *data;
480
481 if (val > 1.0) {
482 val = 1.0 + (val - 1.0) * filter->ratio;
483 } else if (val > threshold) {
484 val = a_p * val * val + b_p * val + c_p;
485 } else if (val < -1.0) {
486 val = -1.0 + (val + 1.0) * filter->ratio;
487 } else if (val < -threshold) {
488 val = a_n * val * val + b_n * val + c_n;
489 }
490 *data++ = (gfloat) val;
491 }
492 }
493
494 static void
gst_audio_dynamic_transform_hard_knee_expander_int(GstAudioDynamic * filter,gint16 * data,guint num_samples)495 gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic * filter,
496 gint16 * data, guint num_samples)
497 {
498 glong val;
499 glong thr_p = filter->threshold * G_MAXINT16;
500 glong thr_n = filter->threshold * G_MININT16;
501 gdouble zero_p, zero_n;
502
503 /* Nothing to do for us here if threshold equals 0.0
504 * or ratio equals 1.0 */
505 if (filter->threshold == 0.0 || filter->ratio == 1.0)
506 return;
507
508 /* zero crossing of our function */
509 if (filter->ratio != 0.0) {
510 zero_p = thr_p - thr_p / filter->ratio;
511 zero_n = thr_n - thr_n / filter->ratio;
512 } else {
513 zero_p = zero_n = 0.0;
514 }
515
516 if (zero_p < 0.0)
517 zero_p = 0.0;
518 if (zero_n > 0.0)
519 zero_n = 0.0;
520
521 for (; num_samples; num_samples--) {
522 val = *data;
523
524 if (val < thr_p && val > zero_p) {
525 val = filter->ratio * val + thr_p * (1 - filter->ratio);
526 } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
527 val = 0;
528 } else if (val > thr_n && val < zero_n) {
529 val = filter->ratio * val + thr_n * (1 - filter->ratio);
530 }
531 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
532 }
533 }
534
535 static void
gst_audio_dynamic_transform_hard_knee_expander_float(GstAudioDynamic * filter,gfloat * data,guint num_samples)536 gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
537 gfloat * data, guint num_samples)
538 {
539 gdouble val, threshold = filter->threshold, zero;
540
541 /* Nothing to do for us here if threshold equals 0.0
542 * or ratio equals 1.0 */
543 if (filter->threshold == 0.0 || filter->ratio == 1.0)
544 return;
545
546 /* zero crossing of our function */
547 if (filter->ratio != 0.0)
548 zero = threshold - threshold / filter->ratio;
549 else
550 zero = 0.0;
551
552 if (zero < 0.0)
553 zero = 0.0;
554
555 for (; num_samples; num_samples--) {
556 val = *data;
557
558 if (val < threshold && val > zero) {
559 val = filter->ratio * val + threshold * (1.0 - filter->ratio);
560 } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
561 val = 0.0;
562 } else if (val > -threshold && val < -zero) {
563 val = filter->ratio * val - threshold * (1.0 - filter->ratio);
564 }
565 *data++ = (gfloat) val;
566 }
567 }
568
569 static void
gst_audio_dynamic_transform_soft_knee_expander_int(GstAudioDynamic * filter,gint16 * data,guint num_samples)570 gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic * filter,
571 gint16 * data, guint num_samples)
572 {
573 glong val;
574 glong thr_p = filter->threshold * G_MAXINT16;
575 glong thr_n = filter->threshold * G_MININT16;
576 gdouble zero_p, zero_n;
577 gdouble a_p, b_p, c_p;
578 gdouble a_n, b_n, c_n;
579 gdouble r2;
580
581 /* Nothing to do for us here if threshold equals 0.0
582 * or ratio equals 1.0 */
583 if (filter->threshold == 0.0 || filter->ratio == 1.0)
584 return;
585
586 /* zero crossing of our function */
587 zero_p = (thr_p * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
588 zero_n = (thr_n * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
589
590 if (zero_p < 0.0)
591 zero_p = 0.0;
592 if (zero_n > 0.0)
593 zero_n = 0.0;
594
595 /* shouldn't happen as this would only happen
596 * with threshold == 0.0 */
597 g_assert (thr_p != 0);
598 g_assert (thr_n != 0);
599
600 /* We build a 2n degree polynomial here for values between
601 * 0 and threshold or 0 and -threshold with:
602 * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
603 * z between 0 and t
604 * =>
605 * a = (1 - r^2) / (4 * t)
606 * b = (1 + r^2) / 2
607 * c = t * (1.0 - b - a*t)
608 * f(x) = ax^2 + bx + c */
609 r2 = filter->ratio * filter->ratio;
610 a_p = (1.0 - r2) / (4.0 * thr_p);
611 b_p = (1.0 + r2) / 2.0;
612 c_p = thr_p * (1.0 - b_p - a_p * thr_p);
613 a_n = (1.0 - r2) / (4.0 * thr_n);
614 b_n = (1.0 + r2) / 2.0;
615 c_n = thr_n * (1.0 - b_n - a_n * thr_n);
616
617 for (; num_samples; num_samples--) {
618 val = *data;
619
620 if (val < thr_p && val > zero_p) {
621 val = a_p * val * val + b_p * val + c_p;
622 } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
623 val = 0;
624 } else if (val > thr_n && val < zero_n) {
625 val = a_n * val * val + b_n * val + c_n;
626 }
627 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
628 }
629 }
630
631 static void
gst_audio_dynamic_transform_soft_knee_expander_float(GstAudioDynamic * filter,gfloat * data,guint num_samples)632 gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
633 gfloat * data, guint num_samples)
634 {
635 gdouble val;
636 gdouble threshold = filter->threshold;
637 gdouble zero;
638 gdouble a_p, b_p, c_p;
639 gdouble a_n, b_n, c_n;
640 gdouble r2;
641
642 /* Nothing to do for us here if threshold equals 0.0
643 * or ratio equals 1.0 */
644 if (filter->threshold == 0.0 || filter->ratio == 1.0)
645 return;
646
647 /* zero crossing of our function */
648 zero = (threshold * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
649
650 if (zero < 0.0)
651 zero = 0.0;
652
653 /* shouldn't happen as this only happens with
654 * threshold == 0.0 */
655 g_assert (threshold != 0.0);
656
657 /* We build a 2n degree polynomial here for values between
658 * 0 and threshold or 0 and -threshold with:
659 * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
660 * z between 0 and t
661 * =>
662 * a = (1 - r^2) / (4 * t)
663 * b = (1 + r^2) / 2
664 * c = t * (1.0 - b - a*t)
665 * f(x) = ax^2 + bx + c */
666 r2 = filter->ratio * filter->ratio;
667 a_p = (1.0 - r2) / (4.0 * threshold);
668 b_p = (1.0 + r2) / 2.0;
669 c_p = threshold * (1.0 - b_p - a_p * threshold);
670 a_n = (1.0 - r2) / (-4.0 * threshold);
671 b_n = (1.0 + r2) / 2.0;
672 c_n = -threshold * (1.0 - b_n + a_n * threshold);
673
674 for (; num_samples; num_samples--) {
675 val = *data;
676
677 if (val < threshold && val > zero) {
678 val = a_p * val * val + b_p * val + c_p;
679 } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
680 val = 0.0;
681 } else if (val > -threshold && val < -zero) {
682 val = a_n * val * val + b_n * val + c_n;
683 }
684 *data++ = (gfloat) val;
685 }
686 }
687
688 /* GstBaseTransform vmethod implementations */
689 static GstFlowReturn
gst_audio_dynamic_transform_ip(GstBaseTransform * base,GstBuffer * buf)690 gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
691 {
692 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
693 guint num_samples;
694 GstClockTime timestamp, stream_time;
695 GstMapInfo map;
696
697 timestamp = GST_BUFFER_TIMESTAMP (buf);
698 stream_time =
699 gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
700
701 GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
702 GST_TIME_ARGS (timestamp));
703
704 if (GST_CLOCK_TIME_IS_VALID (stream_time))
705 gst_object_sync_values (GST_OBJECT (filter), stream_time);
706
707 if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
708 return GST_FLOW_OK;
709
710 gst_buffer_map (buf, &map, GST_MAP_READWRITE);
711 num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);
712
713 filter->process (filter, map.data, num_samples);
714
715 gst_buffer_unmap (buf, &map);
716
717 return GST_FLOW_OK;
718 }
719