diff -r 71e347f905f2 -r 4a7fac7dd34a gst_plugins_good/gst/audiofx/audioinvert.c --- /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 + * Copyright (C) 2006 Stefan Kost + * + * 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. + * + * + * Example launch line + * |[ + * 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 + * ]| + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#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 "); + +/* 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; +}