/* * 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); /* 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_simple (element_class, "Audio inversion", "Filter/Effect/Audio", "Swaps upper and lower half of audio samples", "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); } 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 | G_PARAM_STATIC_STRINGS)); 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; GstClockTime timestamp, stream_time; 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 (G_OBJECT (filter), stream_time); num_samples = GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); 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; }