summaryrefslogtreecommitdiff
path: root/gst/audiofx/audiodynamic.c
blob: e2d87dcb782f4f8a1a0053ffb0e8ecc6c174dee8 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
/* 
 * GStreamer
 * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
 *
 * This library is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Library General Public
 * License as published by the Free Software Foundation; either
 * version 2 of the License, or (at your option) any later version.
 *
 * This library is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 * License along with this library; if not, write to the
 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
 * Boston, MA 02111-1307, USA.
 */

/**
 * SECTION:element-audiodynamic
 *
 * This element can act as a compressor or expander. A compressor changes the
 * amplitude of all samples above a specific threshold with a specific ratio,
 * a expander does the same for all samples below a specific threshold. If
 * soft-knee mode is selected the ratio is applied smoothly.
 *
 * <refsect2>
 * <title>Example launch line</title>
 * |[
 * gst-launch-1.0 audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink
 * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink
 * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
 * ]|
 * </refsect2>
 */

/* TODO: Implement attack and release parameters */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include <gst/audio/audio.h>
#include <gst/audio/gstaudiofilter.h>

#include "audiodynamic.h"

#define GST_CAT_DEFAULT gst_audio_dynamic_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

/* Filter signals and args */
enum
{
  /* FILL ME */
  LAST_SIGNAL
};

enum
{
  PROP_0,
  PROP_CHARACTERISTICS,
  PROP_MODE,
  PROP_THRESHOLD,
  PROP_RATIO
};

#define ALLOWED_CAPS \
    "audio/x-raw,"                                                \
    " format=(string) {"GST_AUDIO_NE(S16)","GST_AUDIO_NE(F32)"}," \
    " rate=(int)[1,MAX],"                                         \
    " channels=(int)[1,MAX],"                                     \
    " layout=(string) {interleaved, non-interleaved}"

G_DEFINE_TYPE (GstAudioDynamic, gst_audio_dynamic, GST_TYPE_AUDIO_FILTER);

static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec);
static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec);

static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,
    const GstAudioInfo * info);
static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,
    GstBuffer * buf);

static void
gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
    gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
    filter, gfloat * data, guint num_samples);
static void
gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
    gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
    filter, gfloat * data, guint num_samples);
static void gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic
    * filter, gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
    gfloat * data, guint num_samples);
static void gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic
    * filter, gint16 * data, guint num_samples);
static void
gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
    gfloat * data, guint num_samples);

static GstAudioDynamicProcessFunc process_functions[] = {
  (GstAudioDynamicProcessFunc)
      gst_audio_dynamic_transform_hard_knee_compressor_int,
  (GstAudioDynamicProcessFunc)
      gst_audio_dynamic_transform_hard_knee_compressor_float,
  (GstAudioDynamicProcessFunc)
      gst_audio_dynamic_transform_soft_knee_compressor_int,
  (GstAudioDynamicProcessFunc)
      gst_audio_dynamic_transform_soft_knee_compressor_float,
  (GstAudioDynamicProcessFunc)
      gst_audio_dynamic_transform_hard_knee_expander_int,
  (GstAudioDynamicProcessFunc)
      gst_audio_dynamic_transform_hard_knee_expander_float,
  (GstAudioDynamicProcessFunc)
      gst_audio_dynamic_transform_soft_knee_expander_int,
  (GstAudioDynamicProcessFunc)
  gst_audio_dynamic_transform_soft_knee_expander_float
};

enum
{
  CHARACTERISTICS_HARD_KNEE = 0,
  CHARACTERISTICS_SOFT_KNEE
};

#define GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS (gst_audio_dynamic_characteristics_get_type ())
static GType
gst_audio_dynamic_characteristics_get_type (void)
{
  static GType gtype = 0;

  if (gtype == 0) {
    static const GEnumValue values[] = {
      {CHARACTERISTICS_HARD_KNEE, "Hard Knee (default)",
          "hard-knee"},
      {CHARACTERISTICS_SOFT_KNEE, "Soft Knee (smooth)",
          "soft-knee"},
      {0, NULL, NULL}
    };

    gtype = g_enum_register_static ("GstAudioDynamicCharacteristics", values);
  }
  return gtype;
}

enum
{
  MODE_COMPRESSOR = 0,
  MODE_EXPANDER
};

#define GST_TYPE_AUDIO_DYNAMIC_MODE (gst_audio_dynamic_mode_get_type ())
static GType
gst_audio_dynamic_mode_get_type (void)
{
  static GType gtype = 0;

  if (gtype == 0) {
    static const GEnumValue values[] = {
      {MODE_COMPRESSOR, "Compressor (default)",
          "compressor"},
      {MODE_EXPANDER, "Expander", "expander"},
      {0, NULL, NULL}
    };

    gtype = g_enum_register_static ("GstAudioDynamicMode", values);
  }
  return gtype;
}

static gboolean
gst_audio_dynamic_set_process_function (GstAudioDynamic * filter,
    const GstAudioInfo * info)
{
  gint func_index;

  if (GST_AUDIO_INFO_FORMAT (info) == GST_AUDIO_FORMAT_UNKNOWN)
    return FALSE;

  func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;
  func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;
  func_index += (GST_AUDIO_INFO_FORMAT (info) == GST_AUDIO_FORMAT_F32) ? 1 : 0;

  if (func_index >= 0 && func_index < 8) {
    filter->process = process_functions[func_index];
    return TRUE;
  }

  return FALSE;
}

/* GObject vmethod implementations */

static void
gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
{
  GObjectClass *gobject_class;
  GstElementClass *gstelement_class;
  GstCaps *caps;

  GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0,
      "audiodynamic element");

  gobject_class = (GObjectClass *) klass;
  gstelement_class = (GstElementClass *) klass;

  gobject_class->set_property = gst_audio_dynamic_set_property;
  gobject_class->get_property = gst_audio_dynamic_get_property;

  g_object_class_install_property (gobject_class, PROP_CHARACTERISTICS,
      g_param_spec_enum ("characteristics", "Characteristics",
          "Selects whether the ratio should be applied smooth (soft-knee) "
          "or hard (hard-knee).",
          GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, CHARACTERISTICS_HARD_KNEE,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_MODE,
      g_param_spec_enum ("mode", "Mode",
          "Selects whether the filter should work on loud samples (compressor) or"
          "quiet samples (expander).",
          GST_TYPE_AUDIO_DYNAMIC_MODE, MODE_COMPRESSOR,
          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_THRESHOLD,
      g_param_spec_float ("threshold", "Threshold",
          "Threshold until the filter is activated", 0.0, 1.0,
          0.0,
          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  g_object_class_install_property (gobject_class, PROP_RATIO,
      g_param_spec_float ("ratio", "Ratio",
          "Ratio that should be applied", 0.0, G_MAXFLOAT,
          1.0,
          G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));

  gst_element_class_set_static_metadata (gstelement_class,
      "Dynamic range controller", "Filter/Effect/Audio",
      "Compressor and Expander", "Sebastian Dröge <slomo@circular-chaos.org>");

  caps = gst_caps_from_string (ALLOWED_CAPS);
  gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
      caps);
  gst_caps_unref (caps);

  GST_AUDIO_FILTER_CLASS (klass)->setup =
      GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);

  GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
      GST_DEBUG_FUNCPTR (gst_audio_dynamic_transform_ip);
  GST_BASE_TRANSFORM_CLASS (klass)->transform_ip_on_passthrough = FALSE;
}

static void
gst_audio_dynamic_init (GstAudioDynamic * filter)
{
  filter->ratio = 1.0;
  filter->threshold = 0.0;
  filter->characteristics = CHARACTERISTICS_HARD_KNEE;
  filter->mode = MODE_COMPRESSOR;
  gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
  gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
}

static void
gst_audio_dynamic_set_property (GObject * object, guint prop_id,
    const GValue * value, GParamSpec * pspec)
{
  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);

  switch (prop_id) {
    case PROP_CHARACTERISTICS:
      filter->characteristics = g_value_get_enum (value);
      gst_audio_dynamic_set_process_function (filter,
          GST_AUDIO_FILTER_INFO (filter));
      break;
    case PROP_MODE:
      filter->mode = g_value_get_enum (value);
      gst_audio_dynamic_set_process_function (filter,
          GST_AUDIO_FILTER_INFO (filter));
      break;
    case PROP_THRESHOLD:
      filter->threshold = g_value_get_float (value);
      break;
    case PROP_RATIO:
      filter->ratio = g_value_get_float (value);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

static void
gst_audio_dynamic_get_property (GObject * object, guint prop_id,
    GValue * value, GParamSpec * pspec)
{
  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);

  switch (prop_id) {
    case PROP_CHARACTERISTICS:
      g_value_set_enum (value, filter->characteristics);
      break;
    case PROP_MODE:
      g_value_set_enum (value, filter->mode);
      break;
    case PROP_THRESHOLD:
      g_value_set_float (value, filter->threshold);
      break;
    case PROP_RATIO:
      g_value_set_float (value, filter->ratio);
      break;
    default:
      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
      break;
  }
}

/* GstAudioFilter vmethod implementations */

static gboolean
gst_audio_dynamic_setup (GstAudioFilter * base, const GstAudioInfo * info)
{
  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
  gboolean ret = TRUE;

  ret = gst_audio_dynamic_set_process_function (filter, info);

  return ret;
}

static void
gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
    gint16 * data, guint num_samples)
{
  glong val;
  glong thr_p = filter->threshold * G_MAXINT16;
  glong thr_n = filter->threshold * G_MININT16;

  /* Nothing to do for us if ratio is 1.0 or if the threshold
   * equals 1.0. */
  if (filter->threshold == 1.0 || filter->ratio == 1.0)
    return;

  for (; num_samples; num_samples--) {
    val = *data;

    if (val > thr_p) {
      val = thr_p + (val - thr_p) * filter->ratio;
    } else if (val < thr_n) {
      val = thr_n + (val - thr_n) * filter->ratio;
    }
    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
  }
}

static void
gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
    filter, gfloat * data, guint num_samples)
{
  gdouble val, threshold = filter->threshold;

  /* Nothing to do for us if ratio == 1.0.
   * As float values can be above 1.0 we have to do something
   * if threshold is greater than 1.0. */
  if (filter->ratio == 1.0)
    return;

  for (; num_samples; num_samples--) {
    val = *data;

    if (val > threshold) {
      val = threshold + (val - threshold) * filter->ratio;
    } else if (val < -threshold) {
      val = -threshold + (val + threshold) * filter->ratio;
    }
    *data++ = (gfloat) val;
  }
}

static void
gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
    gint16 * data, guint num_samples)
{
  glong val;
  glong thr_p = filter->threshold * G_MAXINT16;
  glong thr_n = filter->threshold * G_MININT16;
  gdouble a_p, b_p, c_p;
  gdouble a_n, b_n, c_n;

  /* Nothing to do for us if ratio is 1.0 or if the threshold
   * equals 1.0. */
  if (filter->threshold == 1.0 || filter->ratio == 1.0)
    return;

  /* We build a 2nd degree polynomial here for
   * values greater than threshold or small than
   * -threshold with:
   * f(t) = t, f'(t) = 1, f'(m) = r
   * =>
   * a = (1-r)/(2*(t-m))
   * b = (r*t - m)/(t-m)
   * c = t * (1 - b - a*t)
   * f(x) = ax^2 + bx + c
   */

  /* shouldn't happen because this would only be the case
   * for threshold == 1.0 which we catch above */
  g_assert (thr_p - G_MAXINT16 != 0);
  g_assert (thr_n - G_MININT != 0);

  a_p = (1 - filter->ratio) / (2 * (thr_p - G_MAXINT16));
  b_p = (filter->ratio * thr_p - G_MAXINT16) / (thr_p - G_MAXINT16);
  c_p = thr_p * (1 - b_p - a_p * thr_p);
  a_n = (1 - filter->ratio) / (2 * (thr_n - G_MININT16));
  b_n = (filter->ratio * thr_n - G_MININT16) / (thr_n - G_MININT16);
  c_n = thr_n * (1 - b_n - a_n * thr_n);

  for (; num_samples; num_samples--) {
    val = *data;

    if (val > thr_p) {
      val = a_p * val * val + b_p * val + c_p;
    } else if (val < thr_n) {
      val = a_n * val * val + b_n * val + c_n;
    }
    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
  }
}

static void
gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
    filter, gfloat * data, guint num_samples)
{
  gdouble val;
  gdouble threshold = filter->threshold;
  gdouble a_p, b_p, c_p;
  gdouble a_n, b_n, c_n;

  /* Nothing to do for us if ratio == 1.0.
   * As float values can be above 1.0 we have to do something
   * if threshold is greater than 1.0. */
  if (filter->ratio == 1.0)
    return;

  /* We build a 2nd degree polynomial here for
   * values greater than threshold or small than
   * -threshold with:
   * f(t) = t, f'(t) = 1, f'(m) = r
   * =>
   * a = (1-r)/(2*(t-m))
   * b = (r*t - m)/(t-m)
   * c = t * (1 - b - a*t)
   * f(x) = ax^2 + bx + c
   */

  /* FIXME: If treshold is the same as the maximum
   * we need to raise it a bit to prevent
   * division by zero. */
  if (threshold == 1.0)
    threshold = 1.0 + 0.00001;

  a_p = (1.0 - filter->ratio) / (2.0 * (threshold - 1.0));
  b_p = (filter->ratio * threshold - 1.0) / (threshold - 1.0);
  c_p = threshold * (1.0 - b_p - a_p * threshold);
  a_n = (1.0 - filter->ratio) / (2.0 * (-threshold + 1.0));
  b_n = (-filter->ratio * threshold + 1.0) / (-threshold + 1.0);
  c_n = -threshold * (1.0 - b_n + a_n * threshold);

  for (; num_samples; num_samples--) {
    val = *data;

    if (val > 1.0) {
      val = 1.0 + (val - 1.0) * filter->ratio;
    } else if (val > threshold) {
      val = a_p * val * val + b_p * val + c_p;
    } else if (val < -1.0) {
      val = -1.0 + (val + 1.0) * filter->ratio;
    } else if (val < -threshold) {
      val = a_n * val * val + b_n * val + c_n;
    }
    *data++ = (gfloat) val;
  }
}

static void
gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic * filter,
    gint16 * data, guint num_samples)
{
  glong val;
  glong thr_p = filter->threshold * G_MAXINT16;
  glong thr_n = filter->threshold * G_MININT16;
  gdouble zero_p, zero_n;

  /* Nothing to do for us here if threshold equals 0.0
   * or ratio equals 1.0 */
  if (filter->threshold == 0.0 || filter->ratio == 1.0)
    return;

  /* zero crossing of our function */
  if (filter->ratio != 0.0) {
    zero_p = thr_p - thr_p / filter->ratio;
    zero_n = thr_n - thr_n / filter->ratio;
  } else {
    zero_p = zero_n = 0.0;
  }

  if (zero_p < 0.0)
    zero_p = 0.0;
  if (zero_n > 0.0)
    zero_n = 0.0;

  for (; num_samples; num_samples--) {
    val = *data;

    if (val < thr_p && val > zero_p) {
      val = filter->ratio * val + thr_p * (1 - filter->ratio);
    } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
      val = 0;
    } else if (val > thr_n && val < zero_n) {
      val = filter->ratio * val + thr_n * (1 - filter->ratio);
    }
    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
  }
}

static void
gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
    gfloat * data, guint num_samples)
{
  gdouble val, threshold = filter->threshold, zero;

  /* Nothing to do for us here if threshold equals 0.0
   * or ratio equals 1.0 */
  if (filter->threshold == 0.0 || filter->ratio == 1.0)
    return;

  /* zero crossing of our function */
  if (filter->ratio != 0.0)
    zero = threshold - threshold / filter->ratio;
  else
    zero = 0.0;

  if (zero < 0.0)
    zero = 0.0;

  for (; num_samples; num_samples--) {
    val = *data;

    if (val < threshold && val > zero) {
      val = filter->ratio * val + threshold * (1.0 - filter->ratio);
    } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
      val = 0.0;
    } else if (val > -threshold && val < -zero) {
      val = filter->ratio * val - threshold * (1.0 - filter->ratio);
    }
    *data++ = (gfloat) val;
  }
}

static void
gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic * filter,
    gint16 * data, guint num_samples)
{
  glong val;
  glong thr_p = filter->threshold * G_MAXINT16;
  glong thr_n = filter->threshold * G_MININT16;
  gdouble zero_p, zero_n;
  gdouble a_p, b_p, c_p;
  gdouble a_n, b_n, c_n;
  gdouble r2;

  /* Nothing to do for us here if threshold equals 0.0
   * or ratio equals 1.0 */
  if (filter->threshold == 0.0 || filter->ratio == 1.0)
    return;

  /* zero crossing of our function */
  zero_p = (thr_p * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
  zero_n = (thr_n * (filter->ratio - 1.0)) / (1.0 + filter->ratio);

  if (zero_p < 0.0)
    zero_p = 0.0;
  if (zero_n > 0.0)
    zero_n = 0.0;

  /* shouldn't happen as this would only happen
   * with threshold == 0.0 */
  g_assert (thr_p != 0);
  g_assert (thr_n != 0);

  /* We build a 2n degree polynomial here for values between
   * 0 and threshold or 0 and -threshold with:
   * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
   * z between 0 and t
   * =>
   * a = (1 - r^2) / (4 * t)
   * b = (1 + r^2) / 2
   * c = t * (1.0 - b - a*t)
   * f(x) = ax^2 + bx + c */
  r2 = filter->ratio * filter->ratio;
  a_p = (1.0 - r2) / (4.0 * thr_p);
  b_p = (1.0 + r2) / 2.0;
  c_p = thr_p * (1.0 - b_p - a_p * thr_p);
  a_n = (1.0 - r2) / (4.0 * thr_n);
  b_n = (1.0 + r2) / 2.0;
  c_n = thr_n * (1.0 - b_n - a_n * thr_n);

  for (; num_samples; num_samples--) {
    val = *data;

    if (val < thr_p && val > zero_p) {
      val = a_p * val * val + b_p * val + c_p;
    } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
      val = 0;
    } else if (val > thr_n && val < zero_n) {
      val = a_n * val * val + b_n * val + c_n;
    }
    *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
  }
}

static void
gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
    gfloat * data, guint num_samples)
{
  gdouble val;
  gdouble threshold = filter->threshold;
  gdouble zero;
  gdouble a_p, b_p, c_p;
  gdouble a_n, b_n, c_n;
  gdouble r2;

  /* Nothing to do for us here if threshold equals 0.0
   * or ratio equals 1.0 */
  if (filter->threshold == 0.0 || filter->ratio == 1.0)
    return;

  /* zero crossing of our function */
  zero = (threshold * (filter->ratio - 1.0)) / (1.0 + filter->ratio);

  if (zero < 0.0)
    zero = 0.0;

  /* shouldn't happen as this only happens with
   * threshold == 0.0 */
  g_assert (threshold != 0.0);

  /* We build a 2n degree polynomial here for values between
   * 0 and threshold or 0 and -threshold with:
   * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
   * z between 0 and t
   * =>
   * a = (1 - r^2) / (4 * t)
   * b = (1 + r^2) / 2
   * c = t * (1.0 - b - a*t)
   * f(x) = ax^2 + bx + c */
  r2 = filter->ratio * filter->ratio;
  a_p = (1.0 - r2) / (4.0 * threshold);
  b_p = (1.0 + r2) / 2.0;
  c_p = threshold * (1.0 - b_p - a_p * threshold);
  a_n = (1.0 - r2) / (-4.0 * threshold);
  b_n = (1.0 + r2) / 2.0;
  c_n = -threshold * (1.0 - b_n + a_n * threshold);

  for (; num_samples; num_samples--) {
    val = *data;

    if (val < threshold && val > zero) {
      val = a_p * val * val + b_p * val + c_p;
    } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
      val = 0.0;
    } else if (val > -threshold && val < -zero) {
      val = a_n * val * val + b_n * val + c_n;
    }
    *data++ = (gfloat) val;
  }
}

/* GstBaseTransform vmethod implementations */
static GstFlowReturn
gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
  GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
  guint num_samples;
  GstClockTime timestamp, stream_time;
  GstMapInfo map;

  timestamp = GST_BUFFER_TIMESTAMP (buf);
  stream_time =
      gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);

  GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
      GST_TIME_ARGS (timestamp));

  if (GST_CLOCK_TIME_IS_VALID (stream_time))
    gst_object_sync_values (GST_OBJECT (filter), stream_time);

  if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
    return GST_FLOW_OK;

  gst_buffer_map (buf, &map, GST_MAP_READWRITE);
  num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);

  filter->process (filter, map.data, num_samples);

  gst_buffer_unmap (buf, &map);

  return GST_FLOW_OK;
}