+2007-03-08 Sebastian Dröge <slomo@circular-chaos.org>
+
+ reviewed by: Stefan Kost <ensonic@users.sf.net>
+
+ * gst/audiofx/Makefile.am:
+ * gst/audiofx/audiodynamic.c:
+ (gst_audio_dynamic_characteristics_get_type),
+ (gst_audio_dynamic_mode_get_type),
+ (gst_audio_dynamic_set_process_function),
+ (gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
+ (gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
+ (gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
+ (gst_audio_dynamic_transform_hard_knee_compressor_int),
+ (gst_audio_dynamic_transform_hard_knee_compressor_float),
+ (gst_audio_dynamic_transform_soft_knee_compressor_int),
+ (gst_audio_dynamic_transform_soft_knee_compressor_float),
+ (gst_audio_dynamic_transform_hard_knee_expander_int),
+ (gst_audio_dynamic_transform_hard_knee_expander_float),
+ (gst_audio_dynamic_transform_soft_knee_expander_int),
+ (gst_audio_dynamic_transform_soft_knee_expander_float),
+ (gst_audio_dynamic_transform_ip):
+ * gst/audiofx/audiodynamic.h:
+ * gst/audiofx/audiofx.c: (plugin_init):
+ Add new audiodynamic element which can act as a compressor or
+ expander. Supported are hard-knee and soft-knee operation modes with
+ user-specified ratio and threshold.
+ Attack and release parameters are not yet implemented but will follow.
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ Integrate audiodynamic into the docs.
+ * tests/check/Makefile.am:
+ * tests/check/elements/audiodynamic.c: (setup_dynamic),
+ (cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
+ Add unit test for audiodynamic.
+
2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
$(top_srcdir)/gst/audiofx/audiopanorama.h \
$(top_srcdir)/gst/audiofx/audioinvert.h \
$(top_srcdir)/gst/audiofx/audioamplify.h \
+ $(top_srcdir)/gst/audiofx/audiodynamic.h \
$(top_srcdir)/gst/autodetect/gstautoaudiosink.h \
$(top_srcdir)/gst/autodetect/gstautovideosink.h \
$(top_srcdir)/gst/avi/gstavidemux.h \
<xi:include href="xml/element-audiopanorama.xml" />
<xi:include href="xml/element-audioinvert.xml" />
<xi:include href="xml/element-audioamplify.xml" />
+ <xi:include href="xml/element-audiodynamic.xml" />
<xi:include href="xml/element-autoaudiosink.xml" />
<xi:include href="xml/element-autovideosink.xml" />
<xi:include href="xml/element-avidemux.xml" />
</SECTION>
<SECTION>
+<FILE>element-audiodynamic</FILE>
+GstAudioDynamic
+<TITLE>audiodynamic</TITLE>
+<SUBSECTION Standard>
+GstAudioDynamicClass
+GST_AUDIO_DYNAMIC
+GST_AUDIO_DYNAMIC_CLASS
+</SECTION>
+
+<SECTION>
<FILE>element-autoaudiosink</FILE>
GstAutoAudioSink
<TITLE>autoaudiosink</TITLE>
<DEFAULT>Normal Clipping (default)</DEFAULT>
</ARG>
+<ARG>
+<NAME>GstAudioDynamic::characteristics</NAME>
+<TYPE>GstAudioDynamicCharacteristics</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Characteristics</NICK>
+<BLURB>Selects whether the ratio should be applied smooth (soft-knee) or hard (hard-knee).</BLURB>
+<DEFAULT>Hard Knee (default)</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioDynamic::mode</NAME>
+<TYPE>GstAudioDynamicMode</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Mode</NICK>
+<BLURB>Selects whether the filter should work on loud samples (compressor) orquiet samples (expander).</BLURB>
+<DEFAULT>Compressor (default)</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioDynamic::ratio</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,inf]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Ratio</NICK>
+<BLURB>Ratio that should be applied.</BLURB>
+<DEFAULT>1</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstAudioDynamic::threshold</NAME>
+<TYPE>gfloat</TYPE>
+<RANGE>[0,1]</RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Threshold</NICK>
+<BLURB>Threshold until the filter is activated.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
<author>Sebastian Dröge <slomo@circular-chaos.org></author>
</element>
<element>
+ <name>audiodynamic</name>
+ <longname>AudioDynamic</longname>
+ <class>Filter/Effect/Audio</class>
+ <description>Compressor and Expander</description>
+ <author>Sebastian Dröge <slomo@circular-chaos.org></author>
+ </element>
+ <element>
<name>audioinvert</name>
<longname>AudioInvert</longname>
<class>Filter/Effect/Audio</class>
libgstaudiofx_la_SOURCES = audiofx.c\
audiopanorama.c \
audioinvert.c \
- audioamplify.c
+ audioamplify.c \
+ audiodynamic.c
# flags used to compile this plugin
libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \
# headers we need but don't want installed
noinst_HEADERS = audiopanorama.h \
audioinvert.h \
- audioamplify.h
+ audioamplify.h \
+ audiodynamic.h
+
--- /dev/null
+/*
+ * 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
+ * @short_description: Compressor and Expander
+ *
+ * <refsect2>
+ * 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.
+ * <title>Example launch line</title>
+ * <para>
+ * <programlisting>
+ * gst-launch audiotestsrc wave=saw ! audiodynamic characteristics=soft-knee mode=compressor threshold=0.5 rate=0.5 ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiodynamic characteristics=hard-knee mode=expander threshold=0.2 rate=4.0 ! alsasink
+ * gst-launch audiotestsrc wave=saw ! audioconvert ! audiodynamic ! audioconvert ! alsasink
+ * </programlisting>
+ * </para>
+ * </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 <gst/controller/gstcontroller.h>
+
+#include "audiodynamic.h"
+
+#define GST_CAT_DEFAULT gst_audio_dynamic_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("AudioDynamic",
+ "Filter/Effect/Audio",
+ "Compressor and Expander",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* 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-int," \
+ " depth=(int)16," \
+ " width=(int)16," \
+ " endianness=(int)BYTE_ORDER," \
+ " signed=(bool)TRUE," \
+ " rate=(int)[1,MAX]," \
+ " channels=(int)[1,MAX]; " \
+ "audio/x-raw-float," \
+ " width=(int)32," \
+ " endianness=(int)BYTE_ORDER," \
+ " rate=(int)[1,MAX]," \
+ " channels=(int)[1,MAX]"
+
+#define DEBUG_INIT(bla) \
+ GST_DEBUG_CATEGORY_INIT (gst_audio_dynamic_debug, "audiodynamic", 0, "audiodynamic element");
+
+GST_BOILERPLATE_FULL (GstAudioDynamic, gst_audio_dynamic, GstAudioFilter,
+ GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+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,
+ GstRingBufferSpec * format);
+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)
+{
+ gint func_index;
+
+ func_index = (filter->mode == MODE_COMPRESSOR) ? 0 : 4;
+ func_index += (filter->characteristics == CHARACTERISTICS_HARD_KNEE) ? 0 : 2;
+ func_index += (!filter->is_float) ? 0 : 1;
+
+ 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_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GstCaps *caps;
+
+ gst_element_class_set_details (element_class, &element_details);
+
+ caps = gst_caps_from_string (ALLOWED_CAPS);
+ gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
+ caps);
+ gst_caps_unref (caps);
+}
+
+static void
+gst_audio_dynamic_class_init (GstAudioDynamicClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = (GObjectClass *) 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_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_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_object_class_install_property (gobject_class, PROP_RATIO,
+ g_param_spec_float ("ratio", "Ratio",
+ "Ratio that should be applied", 0.0, G_MAXDOUBLE,
+ 1.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ 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);
+}
+
+static void
+gst_audio_dynamic_init (GstAudioDynamic * filter, GstAudioDynamicClass * klass)
+{
+ filter->ratio = 1.0;
+ filter->threshold = 0.0;
+ filter->characteristics = CHARACTERISTICS_HARD_KNEE;
+ filter->mode = MODE_COMPRESSOR;
+ filter->width = 0;
+ filter->is_float = FALSE;
+ gst_base_transform_set_in_place (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);
+ break;
+ case PROP_MODE:
+ filter->mode = g_value_get_enum (value);
+ gst_audio_dynamic_set_process_function (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, GstRingBufferSpec * format)
+{
+ GstAudioDynamic *filter = GST_AUDIO_DYNAMIC (base);
+ gboolean ret = TRUE;
+
+ filter->width = format->width / 8;
+ filter->is_float = (format->type == GST_BUFTYPE_FLOAT) ? TRUE : FALSE;
+
+ ret = gst_audio_dynamic_set_process_function (filter);
+
+ 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;
+
+ /* 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 */
+ a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_p);
+ b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;
+ c_p = thr_p * (1.0 - b_p - a_p * thr_p);
+ a_n = (1.0 - filter->ratio * filter->ratio) / (4.0 * thr_n);
+ b_n = (1.0 + filter->ratio * filter->ratio) / 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;
+
+ /* 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 */
+ a_p = (1.0 - filter->ratio * filter->ratio) / (4.0 * threshold);
+ b_p = (1.0 + filter->ratio * filter->ratio) / 2.0;
+ c_p = threshold * (1.0 - b_p - a_p * threshold);
+ a_n = (1.0 - filter->ratio * filter->ratio) / (-4.0 * threshold);
+ b_n = (1.0 + filter->ratio * filter->ratio) / 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 = GST_BUFFER_SIZE (buf) / filter->width;
+
+ if (!gst_buffer_is_writable (buf))
+ return GST_FLOW_OK;
+
+ filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}
--- /dev/null
+/*
+ * 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.
+ */
+
+#ifndef __GST_AUDIO_DYNAMIC_H__
+#define __GST_AUDIO_DYNAMIC_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+#include <gst/audio/gstaudiofilter.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_AUDIO_DYNAMIC (gst_audio_dynamic_get_type())
+#define GST_AUDIO_DYNAMIC(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_DYNAMIC,GstAudioDynamic))
+#define GST_IS_AUDIO_DYNAMIC(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_DYNAMIC))
+#define GST_AUDIO_DYNAMIC_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_DYNAMIC,GstAudioDynamicClass))
+#define GST_IS_AUDIO_DYNAMIC_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_DYNAMIC))
+#define GST_AUDIO_DYNAMIC_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_DYNAMIC,GstAudioDynamicClass))
+typedef struct _GstAudioDynamic GstAudioDynamic;
+typedef struct _GstAudioDynamicClass GstAudioDynamicClass;
+
+typedef void (*GstAudioDynamicProcessFunc) (GstAudioDynamic *, guint8 *, guint);
+
+struct _GstAudioDynamic
+{
+ GstAudioFilter audiofilter;
+
+ gfloat degree;
+
+ /* < private > */
+ GstAudioDynamicProcessFunc process;
+ gint width;
+ gboolean is_float;
+ gint characteristics;
+ gint mode;
+ gfloat threshold;
+ gfloat ratio;
+};
+
+struct _GstAudioDynamicClass
+{
+ GstAudioFilterClass parent;
+};
+
+GType gst_audio_dynamic_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_AUDIO_DYNAMIC_H__ */
#include "audiopanorama.h"
#include "audioinvert.h"
#include "audioamplify.h"
+#include "audiodynamic.h"
/* entry point to initialize the plug-in
* initialize the plug-in itself
gst_element_register (plugin, "audioinvert", GST_RANK_NONE,
GST_TYPE_AUDIO_INVERT) &&
gst_element_register (plugin, "audioamplify", GST_RANK_NONE,
- GST_TYPE_AUDIO_AMPLIFY));
+ GST_TYPE_AUDIO_AMPLIFY) &&
+ gst_element_register (plugin, "audiodynamic", GST_RANK_NONE,
+ GST_TYPE_AUDIO_DYNAMIC));
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
elements/audiopanorama \
elements/audioinvert \
elements/audioamplify \
+ elements/audiodynamic \
elements/avimux \
elements/id3demux \
elements/level \
--- /dev/null
+/* GStreamer
+ *
+ * unit test for audiodynamic
+ *
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ *
+ * Greatly based on the audiopanorama unit test
+ * Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
+ *
+ * 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.
+ */
+
+#include <unistd.h>
+
+#include <gst/base/gstbasetransform.h>
+#include <gst/check/gstcheck.h>
+
+gboolean have_eos = FALSE;
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+GstPad *mysrcpad, *mysinkpad;
+
+
+#define DYNAMIC_CAPS_STRING \
+ "audio/x-raw-int, " \
+ "channels = (int) 1, " \
+ "rate = (int) 44100, " \
+ "endianness = (int) BYTE_ORDER, " \
+ "width = (int) 16, " \
+ "depth = (int) 16, " \
+ "signed = (bool) TRUE"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "channels = (int) 1, "
+ "rate = (int) [ 1, MAX ], "
+ "endianness = (int) BYTE_ORDER, "
+ "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")
+ );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "channels = (int) 1, "
+ "rate = (int) [ 1, MAX ], "
+ "endianness = (int) BYTE_ORDER, "
+ "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")
+ );
+
+GstElement *
+setup_dynamic ()
+{
+ GstElement *dynamic;
+
+ GST_DEBUG ("setup_dynamic");
+ dynamic = gst_check_setup_element ("audiodynamic");
+ mysrcpad = gst_check_setup_src_pad (dynamic, &srctemplate, NULL);
+ mysinkpad = gst_check_setup_sink_pad (dynamic, &sinktemplate, NULL);
+ gst_pad_set_active (mysrcpad, TRUE);
+ gst_pad_set_active (mysinkpad, TRUE);
+
+ return dynamic;
+}
+
+void
+cleanup_dynamic (GstElement * dynamic)
+{
+ GST_DEBUG ("cleanup_dynamic");
+
+ g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+ g_list_free (buffers);
+ buffers = NULL;
+
+ gst_pad_set_active (mysrcpad, FALSE);
+ gst_pad_set_active (mysinkpad, FALSE);
+ gst_check_teardown_src_pad (dynamic);
+ gst_check_teardown_sink_pad (dynamic);
+ gst_check_teardown_element (dynamic);
+}
+
+GST_START_TEST (test_passthrough)
+{
+ GstElement *dynamic;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint16 in[6] = { 24576, -16384, 256, -128, 0, -24576 };
+ gint16 *res;
+
+ dynamic = setup_dynamic ();
+ fail_unless (gst_element_set_state (dynamic,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ inbuffer = gst_buffer_new_and_alloc (12);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, 12);
+ fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 12) == 0);
+ caps = gst_caps_from_string (DYNAMIC_CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... but it ends up being collected on the global buffer list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+ GST_INFO
+ ("expected %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld real %+5ld %+5ld %+5ld %+5ld %+5ld %+5ld",
+ in[0], in[1], in[2], in[3], in[4], in[5], res[0], res[1], res[2], res[3],
+ res[4], res[5]);
+ fail_unless (memcmp (GST_BUFFER_DATA (outbuffer), in, 12) == 0);
+
+ /* cleanup */
+ cleanup_dynamic (dynamic);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_compress_hard_50_50)
+{
+ GstElement *dynamic;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint16 in[8] = { -30000, 24576, -16384, 256, -128, 0, -24576, 30000 };
+ gint16 *res;
+
+ dynamic = setup_dynamic ();
+ g_object_set (G_OBJECT (dynamic), "mode", 0, NULL);
+ g_object_set (G_OBJECT (dynamic), "characteristics", 0, NULL);
+ g_object_set (G_OBJECT (dynamic), "ratio", 0.5, NULL);
+ g_object_set (G_OBJECT (dynamic), "threshold", 0.5, NULL);
+ fail_unless (gst_element_set_state (dynamic,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ inbuffer = gst_buffer_new_and_alloc (16);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, 16);
+ fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 16) == 0);
+ caps = gst_caps_from_string (DYNAMIC_CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+
+ fail_unless (res[0] > in[0]);
+ fail_unless (res[1] < in[1]);
+ fail_unless (res[2] == in[2]);
+ fail_unless (res[3] == in[3]);
+ fail_unless (res[4] == in[4]);
+ fail_unless (res[5] == in[5]);
+ fail_unless (res[6] > in[6]);
+ fail_unless (res[7] < in[7]);
+
+ /* cleanup */
+ cleanup_dynamic (dynamic);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_compress_soft_50_50)
+{
+ GstElement *dynamic;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint16 in[8] = { -30000, 24576, -16384, 256, -128, 0, -24576, 30000 };
+ gint16 *res;
+
+ dynamic = setup_dynamic ();
+ g_object_set (G_OBJECT (dynamic), "mode", 0, NULL);
+ g_object_set (G_OBJECT (dynamic), "characteristics", 1, NULL);
+ g_object_set (G_OBJECT (dynamic), "ratio", 0.5, NULL);
+ g_object_set (G_OBJECT (dynamic), "threshold", 0.5, NULL);
+ fail_unless (gst_element_set_state (dynamic,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ inbuffer = gst_buffer_new_and_alloc (16);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, 16);
+ fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 16) == 0);
+ caps = gst_caps_from_string (DYNAMIC_CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+
+ fail_unless (res[0] > in[0]);
+ fail_unless (res[1] < in[1]);
+ fail_unless (res[2] == in[2]);
+ fail_unless (res[3] == in[3]);
+ fail_unless (res[4] == in[4]);
+ fail_unless (res[5] == in[5]);
+ fail_unless (res[6] > in[6]);
+ fail_unless (res[7] < in[7]);
+
+ /* cleanup */
+ cleanup_dynamic (dynamic);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_compress_hard_100_50)
+{
+ GstElement *dynamic;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint16 in[8] = { -30000, 24576, -16384, 256, -128, 0, -24576, 30000 };
+ gint16 *res;
+
+ dynamic = setup_dynamic ();
+ g_object_set (G_OBJECT (dynamic), "mode", 0, NULL);
+ g_object_set (G_OBJECT (dynamic), "characteristics", 0, NULL);
+ g_object_set (G_OBJECT (dynamic), "ratio", 0.5, NULL);
+ g_object_set (G_OBJECT (dynamic), "threshold", 1.0, NULL);
+ fail_unless (gst_element_set_state (dynamic,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ inbuffer = gst_buffer_new_and_alloc (16);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, 16);
+ fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 16) == 0);
+ caps = gst_caps_from_string (DYNAMIC_CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+
+ fail_unless (res[0] == in[0]);
+ fail_unless (res[1] == in[1]);
+ fail_unless (res[2] == in[2]);
+ fail_unless (res[3] == in[3]);
+ fail_unless (res[4] == in[4]);
+ fail_unless (res[5] == in[5]);
+ fail_unless (res[6] == in[6]);
+ fail_unless (res[7] == in[7]);
+
+ /* cleanup */
+ cleanup_dynamic (dynamic);
+}
+
+GST_END_TEST;
+
+
+GST_START_TEST (test_expand_hard_50_200)
+{
+ GstElement *dynamic;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint16 in[8] = { -30000, 24576, -16383, 256, -128, 0, -24576, 30000 };
+ gint16 *res;
+
+ dynamic = setup_dynamic ();
+ g_object_set (G_OBJECT (dynamic), "mode", 1, NULL);
+ g_object_set (G_OBJECT (dynamic), "characteristics", 0, NULL);
+ g_object_set (G_OBJECT (dynamic), "ratio", 2.0, NULL);
+ g_object_set (G_OBJECT (dynamic), "threshold", 0.5, NULL);
+ fail_unless (gst_element_set_state (dynamic,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ inbuffer = gst_buffer_new_and_alloc (16);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, 16);
+ fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 16) == 0);
+ caps = gst_caps_from_string (DYNAMIC_CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+
+ fail_unless (res[0] == in[0]);
+ fail_unless (res[1] == in[1]);
+ fail_unless (res[2] > in[2]);
+ fail_unless (res[3] < in[3]);
+ fail_unless (res[4] > in[4]);
+ fail_unless (res[5] == in[5]);
+ fail_unless (res[6] == in[6]);
+ fail_unless (res[7] == in[7]);
+
+ /* cleanup */
+ cleanup_dynamic (dynamic);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_expand_soft_50_200)
+{
+ GstElement *dynamic;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint16 in[8] = { -30000, 24576, -16383, 256, -128, 0, -24576, 30000 };
+ gint16 *res;
+
+ dynamic = setup_dynamic ();
+ g_object_set (G_OBJECT (dynamic), "mode", 1, NULL);
+ g_object_set (G_OBJECT (dynamic), "characteristics", 1, NULL);
+ g_object_set (G_OBJECT (dynamic), "ratio", 2.0, NULL);
+ g_object_set (G_OBJECT (dynamic), "threshold", 0.5, NULL);
+ fail_unless (gst_element_set_state (dynamic,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ inbuffer = gst_buffer_new_and_alloc (16);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, 16);
+ fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 16) == 0);
+ caps = gst_caps_from_string (DYNAMIC_CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+
+ fail_unless (res[0] == in[0]);
+ fail_unless (res[1] == in[1]);
+ fail_unless (res[2] > in[2]);
+ fail_unless (res[3] < in[3]);
+ fail_unless (res[4] > in[4]);
+ fail_unless (res[5] == in[5]);
+ fail_unless (res[6] == in[6]);
+ fail_unless (res[7] == in[7]);
+
+ /* cleanup */
+ cleanup_dynamic (dynamic);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_expand_hard_0_200)
+{
+ GstElement *dynamic;
+ GstBuffer *inbuffer, *outbuffer;
+ GstCaps *caps;
+ gint16 in[8] = { -30000, 24576, -16383, 256, -128, 0, -24576, 30000 };
+ gint16 *res;
+
+ dynamic = setup_dynamic ();
+ g_object_set (G_OBJECT (dynamic), "mode", 1, NULL);
+ g_object_set (G_OBJECT (dynamic), "characteristics", 0, NULL);
+ g_object_set (G_OBJECT (dynamic), "ratio", 2.0, NULL);
+ g_object_set (G_OBJECT (dynamic), "threshold", 0.0, NULL);
+ fail_unless (gst_element_set_state (dynamic,
+ GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+ "could not set to playing");
+
+ inbuffer = gst_buffer_new_and_alloc (16);
+ memcpy (GST_BUFFER_DATA (inbuffer), in, 16);
+ fail_unless (memcmp (GST_BUFFER_DATA (inbuffer), in, 16) == 0);
+ caps = gst_caps_from_string (DYNAMIC_CAPS_STRING);
+ gst_buffer_set_caps (inbuffer, caps);
+ gst_caps_unref (caps);
+ ASSERT_BUFFER_REFCOUNT (inbuffer, "inbuffer", 1);
+
+ /* pushing gives away my reference ... */
+ fail_unless (gst_pad_push (mysrcpad, inbuffer) == GST_FLOW_OK);
+ /* ... and puts a new buffer on the global list */
+ fail_unless_equals_int (g_list_length (buffers), 1);
+ fail_if ((outbuffer = (GstBuffer *) buffers->data) == NULL);
+
+ res = (gint16 *) GST_BUFFER_DATA (outbuffer);
+
+ fail_unless (res[0] == in[0]);
+ fail_unless (res[1] == in[1]);
+ fail_unless (res[2] == in[2]);
+ fail_unless (res[3] == in[3]);
+ fail_unless (res[4] == in[4]);
+ fail_unless (res[5] == in[5]);
+ fail_unless (res[6] == in[6]);
+ fail_unless (res[7] == in[7]);
+
+ /* cleanup */
+ cleanup_dynamic (dynamic);
+}
+
+GST_END_TEST;
+
+Suite *
+dynamic_suite (void)
+{
+ Suite *s = suite_create ("dynamic");
+ TCase *tc_chain = tcase_create ("general");
+
+ suite_add_tcase (s, tc_chain);
+ tcase_add_test (tc_chain, test_passthrough);
+ tcase_add_test (tc_chain, test_compress_hard_50_50);
+ tcase_add_test (tc_chain, test_compress_soft_50_50);
+ tcase_add_test (tc_chain, test_compress_hard_100_50);
+ tcase_add_test (tc_chain, test_expand_hard_50_200);
+ tcase_add_test (tc_chain, test_expand_soft_50_200);
+ tcase_add_test (tc_chain, test_expand_hard_0_200);
+ return s;
+}
+
+int
+main (int argc, char **argv)
+{
+ int nf;
+
+ Suite *s = dynamic_suite ();
+ SRunner *sr = srunner_create (s);
+
+ gst_check_init (&argc, &argv);
+
+ srunner_run_all (sr, CK_NORMAL);
+ nf = srunner_ntests_failed (sr);
+ srunner_free (sr);
+
+ return nf;
+}
projects / platform / upstream / gstreamer.git / commitdiff