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., 59 Temple Place - Suite 330,

    18  * Boston, MA 02111-1307, USA.

    19  */

    20 

    21 /**

    22  * SECTION:element-audiodynamic

    23  *

    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.

    28  *

    29  * <refsect2>

    30  * <title>Example launch line</title>

    31  * |[

    32  * gst-launch audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink

    33  * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink

    34  * gst-launch audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink

    35  * ]|

    36  * </refsect2>

    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 #include <gst/controller/gstcontroller.h>

    50 

    51 #include "audiodynamic.h"

    52 

    53 #define GST_CAT_DEFAULT gst_audio_dynamic_debug

    54 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);

    55 

    56 static const GstElementDetails element_details =

    57 GST_ELEMENT_DETAILS ("Dynamic range controller",

    58     "Filter/Effect/Audio",

    59     "Compressor and Expander",

    60     "Sebastian Dröge <slomo@circular-chaos.org>");

    61 

    62 /* Filter signals and args */

    63 enum

    64 {

    65   /* FILL ME */

    66   LAST_SIGNAL

    67 };

    68 

    69 enum

    70 {

    71   PROP_0,

    72   PROP_CHARACTERISTICS,

    73   PROP_MODE,

    74   PROP_THRESHOLD,

    75   PROP_RATIO

    76 };

    77 

    78 #define ALLOWED_CAPS \

    79     "audio/x-raw-int,"                                                \

    80     " depth=(int)16,"                                                 \

    81     " width=(int)16,"                                                 \

    82     " endianness=(int)BYTE_ORDER,"                                    \

    83     " signed=(bool)TRUE,"                                             \

    84     " rate=(int)[1,MAX],"                                             \

    85     " channels=(int)[1,MAX]; "                                        \

    86     "audio/x-raw-float,"                                              \

    87     " width=(int)32,"                                                 \

    88     " endianness=(int)BYTE_ORDER,"                                    \

    89     " rate=(int)[1,MAX],"                                             \

    90     " channels=(int)[1,MAX]"

    91 

    92 #define DEBUG_INIT(bla) \

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

    94 

    95 GST_BOILERPLATE_FULL (GstAudioDynamic, gst_audio_dynamic, GstAudioFilter,

    96     GST_TYPE_AUDIO_FILTER, DEBUG_INIT);

    97 

    98 static void gst_audio_dynamic_set_property (GObject * object, guint prop_id,

    99     const GValue * value, GParamSpec * pspec);

   100 static void gst_audio_dynamic_get_property (GObject * object, guint prop_id,

   101     GValue * value, GParamSpec * pspec);

   102 

   103 static gboolean gst_audio_dynamic_setup (GstAudioFilter * filter,

   104     GstRingBufferSpec * format);

   105 static GstFlowReturn gst_audio_dynamic_transform_ip (GstBaseTransform * base,

   106     GstBuffer * buf);

   107 

   108 static void

   109 gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,

   110     gint16 * data, guint num_samples);

   111 static void

   112 gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *

   113     filter, gfloat * data, guint num_samples);

   114 static void

   115 gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,

   116     gint16 * data, guint num_samples);

   117 static void

   118 gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *

   119     filter, gfloat * data, guint num_samples);

   120 static void gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic

   121     * filter, gint16 * data, guint num_samples);

   122 static void

   123 gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,

   124     gfloat * data, guint num_samples);

   125 static void gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic

   126     * filter, gint16 * data, guint num_samples);

   127 static void

   128 gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,

   129     gfloat * data, guint num_samples);

   130 

   131 static GstAudioDynamicProcessFunc process_functions[] = {

   132   (GstAudioDynamicProcessFunc)

   133       gst_audio_dynamic_transform_hard_knee_compressor_int,

   134   (GstAudioDynamicProcessFunc)

   135       gst_audio_dynamic_transform_hard_knee_compressor_float,

   136   (GstAudioDynamicProcessFunc)

   137       gst_audio_dynamic_transform_soft_knee_compressor_int,

   138   (GstAudioDynamicProcessFunc)

   139       gst_audio_dynamic_transform_soft_knee_compressor_float,

   140   (GstAudioDynamicProcessFunc)

   141       gst_audio_dynamic_transform_hard_knee_expander_int,

   142   (GstAudioDynamicProcessFunc)

   143       gst_audio_dynamic_transform_hard_knee_expander_float,

   144   (GstAudioDynamicProcessFunc)

   145       gst_audio_dynamic_transform_soft_knee_expander_int,

   146   (GstAudioDynamicProcessFunc)

   147   gst_audio_dynamic_transform_soft_knee_expander_float

   148 };

   149 

   150 enum

   151 {

   152   CHARACTERISTICS_HARD_KNEE = 0,

   153   CHARACTERISTICS_SOFT_KNEE

   154 };

   155 

   156 #define GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS (gst_audio_dynamic_characteristics_get_type ())

   157 static GType

   158 gst_audio_dynamic_characteristics_get_type (void)

   159 {

   160   static GType gtype = 0;

   161 

   162   if (gtype == 0) {

   163     static const GEnumValue values[] = {

   164       {CHARACTERISTICS_HARD_KNEE, "Hard Knee (default)",

   165           "hard-knee"},

   166       {CHARACTERISTICS_SOFT_KNEE, "Soft Knee (smooth)",

   167           "soft-knee"},

   168       {0, NULL, NULL}

   169     };

   170 

   171     gtype = g_enum_register_static ("GstAudioDynamicCharacteristics", values);

   172   }

   173   return gtype;

   174 }

   175 

   176 enum

   177 {

   178   MODE_COMPRESSOR = 0,

   179   MODE_EXPANDER

   180 };

   181 

   182 #define GST_TYPE_AUDIO_DYNAMIC_MODE (gst_audio_dynamic_mode_get_type ())

   183 static GType

   184 gst_audio_dynamic_mode_get_type (void)

   185 {

   186   static GType gtype = 0;

   187 

   188   if (gtype == 0) {

   189     static const GEnumValue values[] = {

   190       {MODE_COMPRESSOR, "Compressor (default)",

   191           "compressor"},

   192       {MODE_EXPANDER, "Expander", "expander"},

   193       {0, NULL, NULL}

   194     };

   195 

   196     gtype = g_enum_register_static ("GstAudioDynamicMode", values);

   197   }

   198   return gtype;

   199 }

   200 

   201 static gboolean

   202 gst_audio_dynamic_set_process_function (GstAudioDynamic * filter)

   203 {

   204   gint func_index;

   205 

   206   func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;

   207   func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;

   208   func_index +=

   209       (GST_AUDIO_FILTER (filter)->format.type == GST_BUFTYPE_FLOAT) ? 1 : 0;

   210 

   211   if (func_index >= 0 && func_index < 8) {

   212     filter->process = process_functions[func_index];

   213     return TRUE;

   214   }

   215 

   216   return FALSE;

   217 }

   218 

   219 /* GObject vmethod implementations */

   220 

   221 static void

   222 gst_audio_dynamic_base_init (gpointer klass)

   223 {

   224   GstElementClass *element_class = GST_ELEMENT_CLASS (klass);

   225   GstCaps *caps;

   226 

   227   gst_element_class_set_details (element_class, &element_details);

   228 

   229   caps = gst_caps_from_string (ALLOWED_CAPS);

   230   gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),

   231       caps);

   232   gst_caps_unref (caps);

   233 }

   234 

   235 static void

   236 gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)

   237 {

   238   GObjectClass *gobject_class;

   239 

   240   gobject_class = (GObjectClass *) klass;

   241   gobject_class->set_property = gst_audio_dynamic_set_property;

   242   gobject_class->get_property = gst_audio_dynamic_get_property;

   243 

   244   g_object_class_install_property (gobject_class, PROP_CHARACTERISTICS,

   245       g_param_spec_enum ("characteristics", "Characteristics",

   246           "Selects whether the ratio should be applied smooth (soft-knee) "

   247           "or hard (hard-knee).",

   248           GST_TYPE_AUDIO_DYNAMIC_CHARACTERISTICS, CHARACTERISTICS_HARD_KNEE,

   249           G_PARAM_READWRITE));

   250 

   251   g_object_class_install_property (gobject_class, PROP_MODE,

   252       g_param_spec_enum ("mode", "Mode",

   253           "Selects whether the filter should work on loud samples (compressor) or"

   254           "quiet samples (expander).",

   255           GST_TYPE_AUDIO_DYNAMIC_MODE, MODE_COMPRESSOR, G_PARAM_READWRITE));

   256 

   257   g_object_class_install_property (gobject_class, PROP_THRESHOLD,

   258       g_param_spec_float ("threshold", "Threshold",

   259           "Threshold until the filter is activated", 0.0, 1.0,

   260           0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));

   261 

   262   g_object_class_install_property (gobject_class, PROP_RATIO,

   263       g_param_spec_float ("ratio", "Ratio",

   264           "Ratio that should be applied", 0.0, G_MAXFLOAT,

   265           1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));

   266 

   267   GST_AUDIO_FILTER_CLASS (klass)->setup =

   268       GST_DEBUG_FUNCPTR (gst_audio_dynamic_setup);

   269   GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =

   270       GST_DEBUG_FUNCPTR (gst_audio_dynamic_transform_ip);

   271 }

   272 

   273 static void

   274 gst_audio_dynamic_init (GstAudioDynamic * filter, GstAudioDynamicClass * klass)

   275 {

   276   filter->ratio = 1.0;

   277   filter->threshold = 0.0;

   278   filter->characteristics = CHARACTERISTICS_HARD_KNEE;

   279   filter->mode = MODE_COMPRESSOR;

   280   gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);

   281   gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);

   282 }

   283 

   284 static void

   285 gst_audio_dynamic_set_property (GObject * object, guint prop_id,

   286     const GValue * value, GParamSpec * pspec)

   287 {

   288   GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);

   289 

   290   switch (prop_id) {

   291     case PROP_CHARACTERISTICS:

   292       filter->characteristics = g_value_get_enum (value);

   293       gst_audio_dynamic_set_process_function (filter);

   294       break;

   295     case PROP_MODE:

   296       filter->mode = g_value_get_enum (value);

   297       gst_audio_dynamic_set_process_function (filter);

   298       break;

   299     case PROP_THRESHOLD:

   300       filter->threshold = g_value_get_float (value);

   301       break;

   302     case PROP_RATIO:

   303       filter->ratio = g_value_get_float (value);

   304       break;

   305     default:

   306       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

   307       break;

   308   }

   309 }

   310 

   311 static void

   312 gst_audio_dynamic_get_property (GObject * object, guint prop_id,

   313     GValue * value, GParamSpec * pspec)

   314 {

   315   GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (object);

   316 

   317   switch (prop_id) {

   318     case PROP_CHARACTERISTICS:

   319       g_value_set_enum (value, filter->characteristics);

   320       break;

   321     case PROP_MODE:

   322       g_value_set_enum (value, filter->mode);

   323       break;

   324     case PROP_THRESHOLD:

   325       g_value_set_float (value, filter->threshold);

   326       break;

   327     case PROP_RATIO:

   328       g_value_set_float (value, filter->ratio);

   329       break;

   330     default:

   331       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);

   332       break;

   333   }

   334 }

   335 

   336 /* GstAudioFilter vmethod implementations */

   337 

   338 static gboolean

   339 gst_audio_dynamic_setup (GstAudioFilter * base, GstRingBufferSpec * format)

   340 {

   341   GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);

   342   gboolean ret = TRUE;

   343 

   344   ret = gst_audio_dynamic_set_process_function (filter);

   345 

   346   return ret;

   347 }

   348 

   349 static void

   350 gst_audio_dynamic_transform_hard_knee_compressor_int (GstAudioDynamic * filter,

   351     gint16 * data, guint num_samples)

   352 {

   353   glong val;

   354   glong thr_p = filter->threshold * G_MAXINT16;

   355   glong thr_n = filter->threshold * G_MININT16;

   356 

   357   /* Nothing to do for us if ratio is 1.0 or if the threshold

   358    * equals 1.0. */

   359   if (filter->threshold == 1.0 || filter->ratio == 1.0)

   360     return;

   361 

   362   for (; num_samples; num_samples--) {

   363     val = *data;

   364 

   365     if (val > thr_p) {

   366       val = thr_p + (val - thr_p) * filter->ratio;

   367     } else if (val < thr_n) {

   368       val = thr_n + (val - thr_n) * filter->ratio;

   369     }

   370     *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);

   371   }

   372 }

   373 

   374 static void

   375 gst_audio_dynamic_transform_hard_knee_compressor_float (GstAudioDynamic *

   376     filter, gfloat * data, guint num_samples)

   377 {

   378   gdouble val, threshold = filter->threshold;

   379 

   380   /* Nothing to do for us if ratio == 1.0.

   381    * As float values can be above 1.0 we have to do something

   382    * if threshold is greater than 1.0. */

   383   if (filter->ratio == 1.0)

   384     return;

   385 

   386   for (; num_samples; num_samples--) {

   387     val = *data;

   388 

   389     if (val > threshold) {

   390       val = threshold + (val - threshold) * filter->ratio;

   391     } else if (val < -threshold) {

   392       val = -threshold + (val + threshold) * filter->ratio;

   393     }

   394     *data++ = (gfloat) val;

   395   }

   396 }

   397 

   398 static void

   399 gst_audio_dynamic_transform_soft_knee_compressor_int (GstAudioDynamic * filter,

   400     gint16 * data, guint num_samples)

   401 {

   402   glong val;

   403   glong thr_p = filter->threshold * G_MAXINT16;

   404   glong thr_n = filter->threshold * G_MININT16;

   405   gdouble a_p, b_p, c_p;

   406   gdouble a_n, b_n, c_n;

   407 

   408   /* Nothing to do for us if ratio is 1.0 or if the threshold

   409    * equals 1.0. */

   410   if (filter->threshold == 1.0 || filter->ratio == 1.0)

   411     return;

   412 

   413   /* We build a 2nd degree polynomial here for

   414    * values greater than threshold or small than

   415    * -threshold with:

   416    * f(t) = t, f'(t) = 1, f'(m) = r

   417    * =>

   418    * a = (1-r)/(2*(t-m))

   419    * b = (r*t - m)/(t-m)

   420    * c = t * (1 - b - a*t)

   421    * f(x) = ax^2 + bx + c

   422    */

   423 

   424   /* shouldn't happen because this would only be the case

   425    * for threshold == 1.0 which we catch above */

   426   g_assert (thr_p - G_MAXINT16 != 0);

   427   g_assert (thr_n - G_MININT != 0);

   428 

   429   a_p = (1 - filter->ratio) / (2 * (thr_p - G_MAXINT16));

   430   b_p = (filter->ratio * thr_p - G_MAXINT16) / (thr_p - G_MAXINT16);

   431   c_p = thr_p * (1 - b_p - a_p * thr_p);

   432   a_n = (1 - filter->ratio) / (2 * (thr_n - G_MININT16));

   433   b_n = (filter->ratio * thr_n - G_MININT16) / (thr_n - G_MININT16);

   434   c_n = thr_n * (1 - b_n - a_n * thr_n);

   435 

   436   for (; num_samples; num_samples--) {

   437     val = *data;

   438 

   439     if (val > thr_p) {

   440       val = a_p * val * val + b_p * val + c_p;

   441     } else if (val < thr_n) {

   442       val = a_n * val * val + b_n * val + c_n;

   443     }

   444     *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);

   445   }

   446 }

   447 

   448 static void

   449 gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *

   450     filter, gfloat * data, guint num_samples)

   451 {

   452   gdouble val;

   453   gdouble threshold = filter->threshold;

   454   gdouble a_p, b_p, c_p;

   455   gdouble a_n, b_n, c_n;

   456 

   457   /* Nothing to do for us if ratio == 1.0.

   458    * As float values can be above 1.0 we have to do something

   459    * if threshold is greater than 1.0. */

   460   if (filter->ratio == 1.0)

   461     return;

   462 

   463   /* We build a 2nd degree polynomial here for

   464    * values greater than threshold or small than

   465    * -threshold with:

   466    * f(t) = t, f'(t) = 1, f'(m) = r

   467    * =>

   468    * a = (1-r)/(2*(t-m))

   469    * b = (r*t - m)/(t-m)

   470    * c = t * (1 - b - a*t)

   471    * f(x) = ax^2 + bx + c

   472    */

   473 

   474   /* FIXME: If treshold is the same as the maximum

   475    * we need to raise it a bit to prevent

   476    * division by zero. */

   477   if (threshold == 1.0)

   478     threshold = 1.0 + 0.00001;

   479 

   480   a_p = (1.0 - filter->ratio) / (2.0 * (threshold - 1.0));

   481   b_p = (filter->ratio * threshold - 1.0) / (threshold - 1.0);

   482   c_p = threshold * (1.0 - b_p - a_p * threshold);

   483   a_n = (1.0 - filter->ratio) / (2.0 * (-threshold + 1.0));

   484   b_n = (-filter->ratio * threshold + 1.0) / (-threshold + 1.0);

   485   c_n = -threshold * (1.0 - b_n + a_n * threshold);

   486 

   487   for (; num_samples; num_samples--) {

   488     val = *data;

   489 

   490     if (val > 1.0) {

   491       val = 1.0 + (val - 1.0) * filter->ratio;

   492     } else if (val > threshold) {

   493       val = a_p * val * val + b_p * val + c_p;

   494     } else if (val < -1.0) {

   495       val = -1.0 + (val + 1.0) * filter->ratio;

   496     } else if (val < -threshold) {

   497       val = a_n * val * val + b_n * val + c_n;

   498     }

   499     *data++ = (gfloat) val;

   500   }

   501 }

   502 

   503 static void

   504 gst_audio_dynamic_transform_hard_knee_expander_int (GstAudioDynamic * filter,

   505     gint16 * data, guint num_samples)

   506 {

   507   glong val;

   508   glong thr_p = filter->threshold * G_MAXINT16;

   509   glong thr_n = filter->threshold * G_MININT16;

   510   gdouble zero_p, zero_n;

   511 

   512   /* Nothing to do for us here if threshold equals 0.0

   513    * or ratio equals 1.0 */

   514   if (filter->threshold == 0.0 || filter->ratio == 1.0)

   515     return;

   516 

   517   /* zero crossing of our function */

   518   if (filter->ratio != 0.0) {

   519     zero_p = thr_p - thr_p / filter->ratio;

   520     zero_n = thr_n - thr_n / filter->ratio;

   521   } else {

   522     zero_p = zero_n = 0.0;

   523   }

   524 

   525   if (zero_p < 0.0)

   526     zero_p = 0.0;

   527   if (zero_n > 0.0)

   528     zero_n = 0.0;

   529 

   530   for (; num_samples; num_samples--) {

   531     val = *data;

   532 

   533     if (val < thr_p && val > zero_p) {

   534       val = filter->ratio * val + thr_p * (1 - filter->ratio);

   535     } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {

   536       val = 0;

   537     } else if (val > thr_n && val < zero_n) {

   538       val = filter->ratio * val + thr_n * (1 - filter->ratio);

   539     }

   540     *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);

   541   }

   542 }

   543 

   544 static void

   545 gst_audio_dynamic_transform_hard_knee_expander_float (GstAudioDynamic * filter,

   546     gfloat * data, guint num_samples)

   547 {

   548   gdouble val, threshold = filter->threshold, zero;

   549 

   550   /* Nothing to do for us here if threshold equals 0.0

   551    * or ratio equals 1.0 */

   552   if (filter->threshold == 0.0 || filter->ratio == 1.0)

   553     return;

   554 

   555   /* zero crossing of our function */

   556   if (filter->ratio != 0.0)

   557     zero = threshold - threshold / filter->ratio;

   558   else

   559     zero = 0.0;

   560 

   561   if (zero < 0.0)

   562     zero = 0.0;

   563 

   564   for (; num_samples; num_samples--) {

   565     val = *data;

   566 

   567     if (val < threshold && val > zero) {

   568       val = filter->ratio * val + threshold * (1.0 - filter->ratio);

   569     } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {

   570       val = 0.0;

   571     } else if (val > -threshold && val < -zero) {

   572       val = filter->ratio * val - threshold * (1.0 - filter->ratio);

   573     }

   574     *data++ = (gfloat) val;

   575   }

   576 }

   577 

   578 static void

   579 gst_audio_dynamic_transform_soft_knee_expander_int (GstAudioDynamic * filter,

   580     gint16 * data, guint num_samples)

   581 {

   582   glong val;

   583   glong thr_p = filter->threshold * G_MAXINT16;

   584   glong thr_n = filter->threshold * G_MININT16;

   585   gdouble zero_p, zero_n;

   586   gdouble a_p, b_p, c_p;

   587   gdouble a_n, b_n, c_n;

   588 

   589   /* Nothing to do for us here if threshold equals 0.0

   590    * or ratio equals 1.0 */

   591   if (filter->threshold == 0.0 || filter->ratio == 1.0)

   592     return;

   593 

   594   /* zero crossing of our function */

   595   zero_p = (thr_p * (filter->ratio - 1.0)) / (1.0 + filter->ratio);

   596   zero_n = (thr_n * (filter->ratio - 1.0)) / (1.0 + filter->ratio);

   597 

   598   if (zero_p < 0.0)

   599     zero_p = 0.0;

   600   if (zero_n > 0.0)

   601     zero_n = 0.0;

   602 

   603   /* shouldn't happen as this would only happen

   604    * with threshold == 0.0 */

   605   g_assert (thr_p != 0);

   606   g_assert (thr_n != 0);

   607 

   608   /* We build a 2n degree polynomial here for values between

   609    * 0 and threshold or 0 and -threshold with:

   610    * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r

   611    * z between 0 and t

   612    * =>

   613    * a = (1 - r^2) / (4 * t)

   614    * b = (1 + r^2) / 2

   615    * c = t * (1.0 - b - a*t)

   616    * f(x) = ax^2 + bx + c */

   617   a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_p);

   618   b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;

   619   c_p = thr_p * (1.0 - b_p - a_p * thr_p);

   620   a_n = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_n);

   621   b_n = (1.0 + filter->ratio * filter->ratio) / 2.0;

   622   c_n = thr_n * (1.0 - b_n - a_n * thr_n);

   623 

   624   for (; num_samples; num_samples--) {

   625     val = *data;

   626 

   627     if (val < thr_p && val > zero_p) {

   628       val = a_p * val * val + b_p * val + c_p;

   629     } else if ((val <= zero_p && val > 0) || (val >= zero_n && val < 0)) {

   630       val = 0;

   631     } else if (val > thr_n && val < zero_n) {

   632       val = a_n * val * val + b_n * val + c_n;

   633     }

   634     *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);

   635   }

   636 }

   637 

   638 static void

   639 gst_audio_dynamic_transform_soft_knee_expander_float (GstAudioDynamic * filter,

   640     gfloat * data, guint num_samples)

   641 {

   642   gdouble val;

   643   gdouble threshold = filter->threshold;

   644   gdouble zero;

   645   gdouble a_p, b_p, c_p;

   646   gdouble a_n, b_n, c_n;

   647 

   648   /* Nothing to do for us here if threshold equals 0.0

   649    * or ratio equals 1.0 */

   650   if (filter->threshold == 0.0 || filter->ratio == 1.0)

   651     return;

   652 

   653   /* zero crossing of our function */

   654   zero = (threshold * (filter->ratio - 1.0)) / (1.0 + filter->ratio);

   655 

   656   if (zero < 0.0)

   657     zero = 0.0;

   658 

   659   /* shouldn't happen as this only happens with

   660    * threshold == 0.0 */

   661   g_assert (threshold != 0.0);

   662 

   663   /* We build a 2n degree polynomial here for values between

   664    * 0 and threshold or 0 and -threshold with:

   665    * f(t) = t, f'(t) = 1, f(z) = 0, f'(z) = r

   666    * z between 0 and t

   667    * =>

   668    * a = (1 - r^2) / (4 * t)

   669    * b = (1 + r^2) / 2

   670    * c = t * (1.0 - b - a*t)

   671    * f(x) = ax^2 + bx + c */

   672   a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * threshold);

   673   b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;

   674   c_p = threshold * (1.0 - b_p - a_p * threshold);

   675   a_n = (1.0 - filter->ratio * filter->ratio) / (-4.0 * threshold);

   676   b_n = (1.0 + filter->ratio * filter->ratio) / 2.0;

   677   c_n = -threshold * (1.0 - b_n + a_n * threshold);

   678 

   679   for (; num_samples; num_samples--) {

   680     val = *data;

   681 

   682     if (val < threshold && val > zero) {

   683       val = a_p * val * val + b_p * val + c_p;

   684     } else if ((val <= zero && val > 0.0) || (val >= -zero && val < 0.0)) {

   685       val = 0.0;

   686     } else if (val > -threshold && val < -zero) {

   687       val = a_n * val * val + b_n * val + c_n;

   688     }

   689     *data++ = (gfloat) val;

   690   }

   691 }

   692 

   693 /* GstBaseTransform vmethod implementations */

   694 static GstFlowReturn

   695 gst_audio_dynamic_transform_ip (GstBaseTransform * base, GstBuffer * buf)

   696 {

   697   GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);

   698   guint num_samples =

   699       GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);

   700 

   701   if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))

   702     gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));

   703 

   704   if (gst_base_transform_is_passthrough (base) ||

   705       G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))

   706     return GST_FLOW_OK;

   707 

   708   filter->process (filter, GST_BUFFER_DATA (buf), num_samples);

   709 

   710   return GST_FLOW_OK;

   711 }