| /* |
| * 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., 51 Franklin St, Fifth Floor, |
| * Boston, MA 02110-1301, 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 ratio=0.5 ! alsasink |
| * gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 ratio=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 const 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 void |
| gst_audio_dynamic_set_process_function (GstAudioDynamic * filter, |
| const GstAudioInfo * info) |
| { |
| gint func_index; |
| |
| 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; |
| |
| g_assert (func_index >= 0 && func_index < G_N_ELEMENTS (process_functions)); |
| |
| filter->process = process_functions[func_index]; |
| } |
| |
| /* 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); |
| |
| gst_audio_dynamic_set_process_function (filter, info); |
| return TRUE; |
| } |
| |
| 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; |
| } |