--- /dev/null Thu Jan 01 00:00:00 1970 +0000
+++ b/gst_plugins_good/gst/audiofx/audioinvert.c Fri Apr 16 15:15:52 2010 +0300
@@ -0,0 +1,252 @@
+/*
+ * GStreamer
+ * Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
+ * 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.
+ */
+
+/**
+ * SECTION:element-audioinvert
+ *
+ * Swaps upper and lower half of audio samples. Mixing an inverted sample on top of
+ * the original with a slight delay can produce effects that sound like resonance.
+ * Creating a stereo sample from a mono source, with one channel inverted produces wide-stereo sounds.
+ *
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch audiotestsrc wave=saw ! audioinvert invert=0.4 ! alsasink
+ * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert invert=0.4 ! alsasink
+ * gst-launch audiotestsrc wave=saw ! audioconvert ! audioinvert invert=0.4 ! audioconvert ! alsasink
+ * ]|
+ * </refsect2>
+ */
+
+#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 "audioinvert.h"
+
+#define GST_CAT_DEFAULT gst_audio_invert_debug
+GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
+
+static const GstElementDetails element_details =
+GST_ELEMENT_DETAILS ("Audio inversion",
+ "Filter/Effect/Audio",
+ "Swaps upper and lower half of audio samples",
+ "Sebastian Dröge <slomo@circular-chaos.org>");
+
+/* Filter signals and args */
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+ PROP_DEGREE
+};
+
+#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_invert_debug, "audioinvert", 0, "audioinvert element");
+
+GST_BOILERPLATE_FULL (GstAudioInvert, gst_audio_invert, GstAudioFilter,
+ GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
+
+static void gst_audio_invert_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_audio_invert_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+static gboolean gst_audio_invert_setup (GstAudioFilter * filter,
+ GstRingBufferSpec * format);
+static GstFlowReturn gst_audio_invert_transform_ip (GstBaseTransform * base,
+ GstBuffer * buf);
+
+static void gst_audio_invert_transform_int (GstAudioInvert * filter,
+ gint16 * data, guint num_samples);
+static void gst_audio_invert_transform_float (GstAudioInvert * filter,
+ gfloat * data, guint num_samples);
+
+/* GObject vmethod implementations */
+
+static void
+gst_audio_invert_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_invert_class_init (GstAudioInvertClass * klass)
+{
+ GObjectClass *gobject_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gobject_class->set_property = gst_audio_invert_set_property;
+ gobject_class->get_property = gst_audio_invert_get_property;
+
+ g_object_class_install_property (gobject_class, PROP_DEGREE,
+ g_param_spec_float ("degree", "Degree",
+ "Degree of inversion", 0.0, 1.0,
+ 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
+
+ GST_AUDIO_FILTER_CLASS (klass)->setup =
+ GST_DEBUG_FUNCPTR (gst_audio_invert_setup);
+ GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
+ GST_DEBUG_FUNCPTR (gst_audio_invert_transform_ip);
+}
+
+static void
+gst_audio_invert_init (GstAudioInvert * filter, GstAudioInvertClass * klass)
+{
+ filter->degree = 0.0;
+ 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_invert_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstAudioInvert *filter = GST_AUDIO_INVERT (object);
+
+ switch (prop_id) {
+ case PROP_DEGREE:
+ filter->degree = g_value_get_float (value);
+ gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
+ filter->degree == 0.0);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_audio_invert_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstAudioInvert *filter = GST_AUDIO_INVERT (object);
+
+ switch (prop_id) {
+ case PROP_DEGREE:
+ g_value_set_float (value, filter->degree);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+/* GstAudioFilter vmethod implementations */
+
+static gboolean
+gst_audio_invert_setup (GstAudioFilter * base, GstRingBufferSpec * format)
+{
+ GstAudioInvert *filter = GST_AUDIO_INVERT (base);
+ gboolean ret = TRUE;
+
+ if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
+ filter->process = (GstAudioInvertProcessFunc)
+ gst_audio_invert_transform_float;
+ else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
+ filter->process = (GstAudioInvertProcessFunc)
+ gst_audio_invert_transform_int;
+ else
+ ret = FALSE;
+
+ return ret;
+}
+
+static void
+gst_audio_invert_transform_int (GstAudioInvert * filter,
+ gint16 * data, guint num_samples)
+{
+ gint i;
+ gfloat dry = 1.0 - filter->degree;
+ glong val;
+
+ for (i = 0; i < num_samples; i++) {
+ val = (*data) * dry + (-1 - (*data)) * filter->degree;
+ *data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
+ }
+}
+
+static void
+gst_audio_invert_transform_float (GstAudioInvert * filter,
+ gfloat * data, guint num_samples)
+{
+ gint i;
+ gfloat dry = 1.0 - filter->degree;
+ glong val;
+
+ for (i = 0; i < num_samples; i++) {
+ val = (*data) * dry - (*data) * filter->degree;
+ *data++ = val;
+ }
+}
+
+/* GstBaseTransform vmethod implementations */
+static GstFlowReturn
+gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+ GstAudioInvert *filter = GST_AUDIO_INVERT (base);
+ guint num_samples =
+ GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
+
+ if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf)))
+ gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf));
+
+ if (gst_base_transform_is_passthrough (base) ||
+ G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
+ return GST_FLOW_OK;
+
+ filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
+
+ return GST_FLOW_OK;
+}