3 * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
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.
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.
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.
22 * SECTION:element-audiodynamic
24 * This element can act as a compressor or expander. A compressor changes the
25 * amplitude of all samples above a specific threshold with a specific ratio,
26 * a expander does the same for all samples below a specific threshold. If
27 * soft-knee mode is selected the ratio is applied smoothly.
30 * <title>Example launch line</title>
32 * gst-launch-1.0 audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink
33 * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink
34 * gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
39 /* TODO: Implement attack and release parameters */
46 #include <gst/base/gstbasetransform.h>
47 #include <gst/audio/audio.h>
48 #include <gst/audio/gstaudiofilter.h>
50 #include "audiodynamic.h"
52 #define GST_CAT_DEFAULT gst_audio_dynamic_debug
53 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
55 /* Filter signals and args */
71 #define ALLOWED_CAPS \
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}"
78 G_DEFINE_TYPE (GstAudioDynamic, gst_audio_dynamic, GST_TYPE_AUDIO_FILTER);
80 static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,
81 const GValue * value, GParamSpec * pspec);
82 static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,
83 GValue * value, GParamSpec * pspec);
85 static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,
86 const GstAudioInfo * info);
87 static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,
91 gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
92 gint16 * data, guint num_samples);
94 gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
95 filter, gfloat * data, guint num_samples);
97 gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
98 gint16 * data, guint num_samples);
100 gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
101 filter, gfloat * data, guint num_samples);
102 static void gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic
103 * filter, gint16 * data, guint num_samples);
105 gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
106 gfloat * data, guint num_samples);
107 static void gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic
108 * filter, gint16 * data, guint num_samples);
110 gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
111 gfloat * data, guint num_samples);
113 static GstAudioDynamicProcessFunc process_functions[] = {
114 (GstAudioDynamicProcessFunc)
115 gst_audio_dynamic_transform_hard_knee_compressor_int,
116 (GstAudioDynamicProcessFunc)
117 gst_audio_dynamic_transform_hard_knee_compressor_float,
118 (GstAudioDynamicProcessFunc)
119 gst_audio_dynamic_transform_soft_knee_compressor_int,
120 (GstAudioDynamicProcessFunc)
121 gst_audio_dynamic_transform_soft_knee_compressor_float,
122 (GstAudioDynamicProcessFunc)
123 gst_audio_dynamic_transform_hard_knee_expander_int,
124 (GstAudioDynamicProcessFunc)
125 gst_audio_dynamic_transform_hard_knee_expander_float,
126 (GstAudioDynamicProcessFunc)
127 gst_audio_dynamic_transform_soft_knee_expander_int,
128 (GstAudioDynamicProcessFunc)
129 gst_audio_dynamic_transform_soft_knee_expander_float
134 CHARACTERISTICS_HARD_KNEE = 0,
135 CHARACTERISTICS_SOFT_KNEE
138 #define GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS (gst_audio_dynamic_characteristics_get_type ())
140 gst_audio_dynamic_characteristics_get_type (void)
142 static GType gtype = 0;
145 static const GEnumValue values[] = {
146 {CHARACTERISTICS_HARD_KNEE, "Hard Knee (default)",
148 {CHARACTERISTICS_SOFT_KNEE, "Soft Knee (smooth)",
153 gtype = g_enum_register_static ("GstAudioDynamicCharacteristics", values);
164 #define GST_TYPE_AUDIO_DYNAMIC_MODE (gst_audio_dynamic_mode_get_type ())
166 gst_audio_dynamic_mode_get_type (void)
168 static GType gtype = 0;
171 static const GEnumValue values[] = {
172 {MODE_COMPRESSOR, "Compressor (default)",
174 {MODE_EXPANDER, "Expander", "expander"},
178 gtype = g_enum_register_static ("GstAudioDynamicMode", values);
184 gst_audio_dynamic_set_process_function (GstAudioDynamic * filter,
185 const GstAudioInfo * info)
189 if (GST_AUDIO_INFO_FORMAT (info) == GST_AUDIO_FORMAT_UNKNOWN)
192 func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;
193 func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;
194 func_index += (GST_AUDIO_INFO_FORMAT (info) == GST_AUDIO_FORMAT_F32) ? 1 : 0;
196 if (func_index >= 0 && func_index < 8) {
197 filter->process = process_functions[func_index];
204 /* GObject vmethod implementations */
207 gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
209 GObjectClass *gobject_class;
210 GstElementClass *gstelement_class;
213 GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0,
214 "audiodynamic element");
216 gobject_class = (GObjectClass *) klass;
217 gstelement_class = (GstElementClass *) klass;
219 gobject_class->set_property = gst_audio_dynamic_set_property;
220 gobject_class->get_property = gst_audio_dynamic_get_property;
222 g_object_class_install_property (gobject_class, PROP_CHARACTERISTICS,
223 g_param_spec_enum ("characteristics", "Characteristics",
224 "Selects whether the ratio should be applied smooth (soft-knee) "
225 "or hard (hard-knee).",
226 GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, CHARACTERISTICS_HARD_KNEE,
227 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
229 g_object_class_install_property (gobject_class, PROP_MODE,
230 g_param_spec_enum ("mode", "Mode",
231 "Selects whether the filter should work on loud samples (compressor) or"
232 "quiet samples (expander).",
233 GST_TYPE_AUDIO_DYNAMIC_MODE, MODE_COMPRESSOR,
234 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
236 g_object_class_install_property (gobject_class, PROP_THRESHOLD,
237 g_param_spec_float ("threshold", "Threshold",
238 "Threshold until the filter is activated", 0.0, 1.0,
240 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
242 g_object_class_install_property (gobject_class, PROP_RATIO,
243 g_param_spec_float ("ratio", "Ratio",
244 "Ratio that should be applied", 0.0, G_MAXFLOAT,
246 G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE | G_PARAM_STATIC_STRINGS));
248 gst_element_class_set_static_metadata (gstelement_class,
249 "Dynamic range controller", "Filter/Effect/Audio",
250 "Compressor and Expander", "Sebastian Dröge <slomo@circular-chaos.org>");
252 caps = gst_caps_from_string (ALLOWED_CAPS);
253 gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
255 gst_caps_unref (caps);
257 GST_AUDIO_FILTER_CLASS (klass)->setup =
258 GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);
260 GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
261 GST_DEBUG_FUNCPTR (gst_audio_dynamic_transform_ip);
262 GST_BASE_TRANSFORM_CLASS (klass)->transform_ip_on_passthrough = FALSE;
266 gst_audio_dynamic_init (GstAudioDynamic * filter)
269 filter->threshold = 0.0;
270 filter->characteristics = CHARACTERISTICS_HARD_KNEE;
271 filter->mode = MODE_COMPRESSOR;
272 gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
273 gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
277 gst_audio_dynamic_set_property (GObject * object, guint prop_id,
278 const GValue * value, GParamSpec * pspec)
280 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
283 case PROP_CHARACTERISTICS:
284 filter->characteristics = g_value_get_enum (value);
285 gst_audio_dynamic_set_process_function (filter,
286 GST_AUDIO_FILTER_INFO (filter));
289 filter->mode = g_value_get_enum (value);
290 gst_audio_dynamic_set_process_function (filter,
291 GST_AUDIO_FILTER_INFO (filter));
294 filter->threshold = g_value_get_float (value);
297 filter->ratio = g_value_get_float (value);
300 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
306 gst_audio_dynamic_get_property (GObject * object, guint prop_id,
307 GValue * value, GParamSpec * pspec)
309 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);
312 case PROP_CHARACTERISTICS:
313 g_value_set_enum (value, filter->characteristics);
316 g_value_set_enum (value, filter->mode);
319 g_value_set_float (value, filter->threshold);
322 g_value_set_float (value, filter->ratio);
325 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
330 /* GstAudioFilter vmethod implementations */
333 gst_audio_dynamic_setup (GstAudioFilter * base, const GstAudioInfo * info)
335 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
338 ret = gst_audio_dynamic_set_process_function (filter, info);
344 gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,
345 gint16 * data, guint num_samples)
348 glong thr_p = filter->threshold * G_MAXINT16;
349 glong thr_n = filter->threshold * G_MININT16;
351 /* Nothing to do for us if ratio is 1.0 or if the threshold
353 if (filter->threshold == 1.0 || filter->ratio == 1.0)
356 for (; num_samples; num_samples--) {
360 val = thr_p + (val - thr_p) * filter->ratio;
361 } else if (val < thr_n) {
362 val = thr_n + (val - thr_n) * filter->ratio;
364 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
369 gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *
370 filter, gfloat * data, guint num_samples)
372 gdouble val, threshold = filter->threshold;
374 /* Nothing to do for us if ratio == 1.0.
375 * As float values can be above 1.0 we have to do something
376 * if threshold is greater than 1.0. */
377 if (filter->ratio == 1.0)
380 for (; num_samples; num_samples--) {
383 if (val > threshold) {
384 val = threshold + (val - threshold) * filter->ratio;
385 } else if (val < -threshold) {
386 val = -threshold + (val + threshold) * filter->ratio;
388 *data++ = (gfloat) val;
393 gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,
394 gint16 * data, guint num_samples)
397 glong thr_p = filter->threshold * G_MAXINT16;
398 glong thr_n = filter->threshold * G_MININT16;
399 gdouble a_p, b_p, c_p;
400 gdouble a_n, b_n, c_n;
402 /* Nothing to do for us if ratio is 1.0 or if the threshold
404 if (filter->threshold == 1.0 || filter->ratio == 1.0)
407 /* We build a 2nd degree polynomial here for
408 * values greater than threshold or small than
410 * f(t) = t, f'(t) = 1, f'(m) = r
412 * a = (1-r)/(2*(t-m))
413 * b = (r*t - m)/(t-m)
414 * c = t * (1 - b - a*t)
415 * f(x) = ax^2 + bx + c
418 /* shouldn't happen because this would only be the case
419 * for threshold == 1.0 which we catch above */
420 g_assert (thr_p - G_MAXINT16 != 0);
421 g_assert (thr_n - G_MININT != 0);
423 a_p = (1 - filter->ratio) / (2 * (thr_p - G_MAXINT16));
424 b_p = (filter->ratio * thr_p - G_MAXINT16) / (thr_p - G_MAXINT16);
425 c_p = thr_p * (1 - b_p - a_p * thr_p);
426 a_n = (1 - filter->ratio) / (2 * (thr_n - G_MININT16));
427 b_n = (filter->ratio * thr_n - G_MININT16) / (thr_n - G_MININT16);
428 c_n = thr_n * (1 - b_n - a_n * thr_n);
430 for (; num_samples; num_samples--) {
434 val = a_p * val * val + b_p * val + c_p;
435 } else if (val < thr_n) {
436 val = a_n * val * val + b_n * val + c_n;
438 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
443 gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
444 filter, gfloat * data, guint num_samples)
447 gdouble threshold = filter->threshold;
448 gdouble a_p, b_p, c_p;
449 gdouble a_n, b_n, c_n;
451 /* Nothing to do for us if ratio == 1.0.
452 * As float values can be above 1.0 we have to do something
453 * if threshold is greater than 1.0. */
454 if (filter->ratio == 1.0)
457 /* We build a 2nd degree polynomial here for
458 * values greater than threshold or small than
460 * f(t) = t, f'(t) = 1, f'(m) = r
462 * a = (1-r)/(2*(t-m))
463 * b = (r*t - m)/(t-m)
464 * c = t * (1 - b - a*t)
465 * f(x) = ax^2 + bx + c
468 /* FIXME: If treshold is the same as the maximum
469 * we need to raise it a bit to prevent
470 * division by zero. */
471 if (threshold == 1.0)
472 threshold = 1.0 + 0.00001;
474 a_p = (1.0 - filter->ratio) / (2.0 * (threshold - 1.0));
475 b_p = (filter->ratio * threshold - 1.0) / (threshold - 1.0);
476 c_p = threshold * (1.0 - b_p - a_p * threshold);
477 a_n = (1.0 - filter->ratio) / (2.0 * (-threshold + 1.0));
478 b_n = (-filter->ratio * threshold + 1.0) / (-threshold + 1.0);
479 c_n = -threshold * (1.0 - b_n + a_n * threshold);
481 for (; num_samples; num_samples--) {
485 val = 1.0 + (val - 1.0) * filter->ratio;
486 } else if (val > threshold) {
487 val = a_p * val * val + b_p * val + c_p;
488 } else if (val < -1.0) {
489 val = -1.0 + (val + 1.0) * filter->ratio;
490 } else if (val < -threshold) {
491 val = a_n * val * val + b_n * val + c_n;
493 *data++ = (gfloat) val;
498 gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic * filter,
499 gint16 * data, guint num_samples)
502 glong thr_p = filter->threshold * G_MAXINT16;
503 glong thr_n = filter->threshold * G_MININT16;
504 gdouble zero_p, zero_n;
506 /* Nothing to do for us here if threshold equals 0.0
507 * or ratio equals 1.0 */
508 if (filter->threshold == 0.0 || filter->ratio == 1.0)
511 /* zero crossing of our function */
512 if (filter->ratio != 0.0) {
513 zero_p = thr_p - thr_p / filter->ratio;
514 zero_n = thr_n - thr_n / filter->ratio;
516 zero_p = zero_n = 0.0;
524 for (; num_samples; num_samples--) {
527 if (val < thr_p && val > zero_p) {
528 val = filter->ratio * val + thr_p * (1 - filter->ratio);
529 } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
531 } else if (val > thr_n && val < zero_n) {
532 val = filter->ratio * val + thr_n * (1 - filter->ratio);
534 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
539 gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,
540 gfloat * data, guint num_samples)
542 gdouble val, threshold = filter->threshold, zero;
544 /* Nothing to do for us here if threshold equals 0.0
545 * or ratio equals 1.0 */
546 if (filter->threshold == 0.0 || filter->ratio == 1.0)
549 /* zero crossing of our function */
550 if (filter->ratio != 0.0)
551 zero = threshold - threshold / filter->ratio;
558 for (; num_samples; num_samples--) {
561 if (val < threshold && val > zero) {
562 val = filter->ratio * val + threshold * (1.0 - filter->ratio);
563 } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
565 } else if (val > -threshold && val < -zero) {
566 val = filter->ratio * val - threshold * (1.0 - filter->ratio);
568 *data++ = (gfloat) val;
573 gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic * filter,
574 gint16 * data, guint num_samples)
577 glong thr_p = filter->threshold * G_MAXINT16;
578 glong thr_n = filter->threshold * G_MININT16;
579 gdouble zero_p, zero_n;
580 gdouble a_p, b_p, c_p;
581 gdouble a_n, b_n, c_n;
584 /* Nothing to do for us here if threshold equals 0.0
585 * or ratio equals 1.0 */
586 if (filter->threshold == 0.0 || filter->ratio == 1.0)
589 /* zero crossing of our function */
590 zero_p = (thr_p * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
591 zero_n = (thr_n * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
598 /* shouldn't happen as this would only happen
599 * with threshold == 0.0 */
600 g_assert (thr_p != 0);
601 g_assert (thr_n != 0);
603 /* We build a 2n degree polynomial here for values between
604 * 0 and threshold or 0 and -threshold with:
605 * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
608 * a = (1 - r^2) / (4 * t)
610 * c = t * (1.0 - b - a*t)
611 * f(x) = ax^2 + bx + c */
612 r2 = filter->ratio * filter->ratio;
613 a_p = (1.0 - r2) / (4.0 * thr_p);
614 b_p = (1.0 + r2) / 2.0;
615 c_p = thr_p * (1.0 - b_p - a_p * thr_p);
616 a_n = (1.0 - r2) / (4.0 * thr_n);
617 b_n = (1.0 + r2) / 2.0;
618 c_n = thr_n * (1.0 - b_n - a_n * thr_n);
620 for (; num_samples; num_samples--) {
623 if (val < thr_p && val > zero_p) {
624 val = a_p * val * val + b_p * val + c_p;
625 } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {
627 } else if (val > thr_n && val < zero_n) {
628 val = a_n * val * val + b_n * val + c_n;
630 *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
635 gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,
636 gfloat * data, guint num_samples)
639 gdouble threshold = filter->threshold;
641 gdouble a_p, b_p, c_p;
642 gdouble a_n, b_n, c_n;
645 /* Nothing to do for us here if threshold equals 0.0
646 * or ratio equals 1.0 */
647 if (filter->threshold == 0.0 || filter->ratio == 1.0)
650 /* zero crossing of our function */
651 zero = (threshold * (filter->ratio - 1.0)) / (1.0 + filter->ratio);
656 /* shouldn't happen as this only happens with
657 * threshold == 0.0 */
658 g_assert (threshold != 0.0);
660 /* We build a 2n degree polynomial here for values between
661 * 0 and threshold or 0 and -threshold with:
662 * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r
665 * a = (1 - r^2) / (4 * t)
667 * c = t * (1.0 - b - a*t)
668 * f(x) = ax^2 + bx + c */
669 r2 = filter->ratio * filter->ratio;
670 a_p = (1.0 - r2) / (4.0 * threshold);
671 b_p = (1.0 + r2) / 2.0;
672 c_p = threshold * (1.0 - b_p - a_p * threshold);
673 a_n = (1.0 - r2) / (-4.0 * threshold);
674 b_n = (1.0 + r2) / 2.0;
675 c_n = -threshold * (1.0 - b_n + a_n * threshold);
677 for (; num_samples; num_samples--) {
680 if (val < threshold && val > zero) {
681 val = a_p * val * val + b_p * val + c_p;
682 } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {
684 } else if (val > -threshold && val < -zero) {
685 val = a_n * val * val + b_n * val + c_n;
687 *data++ = (gfloat) val;
691 /* GstBaseTransform vmethod implementations */
693 gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)
695 GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
697 GstClockTime timestamp, stream_time;
700 timestamp = GST_BUFFER_TIMESTAMP (buf);
702 gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
704 GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
705 GST_TIME_ARGS (timestamp));
707 if (GST_CLOCK_TIME_IS_VALID (stream_time))
708 gst_object_sync_values (GST_OBJECT (filter), stream_time);
710 if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
713 gst_buffer_map (buf, &map, GST_MAP_READWRITE);
714 num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);
716 filter->process (filter, map.data, num_samples);
718 gst_buffer_unmap (buf, &map);