/* * 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 * @short_description: Swaps upper and lower half of audio samples * * * 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 "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 ("AudioInvert", "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 }; static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], " "endianness = (int) BYTE_ORDER, " "width = (int) 32; " "audio/x-raw-int, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], " "endianness = (int) BYTE_ORDER, " "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true") ); static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX], " "endianness = (int) BYTE_ORDER, " "width = (int) 32; " "audio/x-raw-int, " "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ], " "endianness = (int) BYTE_ORDER, " "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) true") ); #define DEBUG_INIT(bla) \ GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0, "audioinvert element"); GST_BOILERPLATE_FULL (GstAudioInvert, gst_audio_invert, GstBaseTransform, GST_TYPE_BASE_TRANSFORM, 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_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps); 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); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_template)); gst_element_class_set_details (element_class, &element_details); } 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_BASE_TRANSFORM_CLASS (klass)->set_caps = GST_DEBUG_FUNCPTR (gst_audio_invert_set_caps); 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; filter->width = 0; gst_base_transform_set_in_place (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; } } /* GstBaseTransform vmethod implementations */ static gboolean gst_audio_invert_set_caps (GstBaseTransform * base, GstCaps * incaps, GstCaps * outcaps) { GstAudioInvert *filter = GST_AUDIO_INVERT (base); const GstStructure *structure; gboolean ret; gint width; const gchar *fmt; /*GST_INFO ("incaps are %" GST_PTR_FORMAT, incaps); */ structure = gst_caps_get_structure (incaps, 0); ret = gst_structure_get_int (structure, "width", &width); if (!ret) goto no_width; filter->width = width / 8; fmt = gst_structure_get_name (structure); if (!strcmp (fmt, "audio/x-raw-int")) filter->process = (GstAudioInvertProcessFunc) gst_audio_invert_transform_int; else filter->process = (GstAudioInvertProcessFunc) gst_audio_invert_transform_float; return TRUE; no_width: GST_DEBUG ("no width in caps"); return FALSE; } 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; } } /* this function does the actual processing */ static GstFlowReturn gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf) { GstAudioInvert *filter = GST_AUDIO_INVERT (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; }