gstreamer/gst/audiofx/audioinvert.c
Sebastian Dröge 447ae144c2 gst/audiofx/: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can b...
Original commit message from CVS:
reviewed by: Stefan Kost  <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_init),
(gst_audio_invert_set_property), (gst_audio_invert_get_property),
(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
(gst_audio_invert_transform_float),
(gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Add new audiofx element "audioinvert". This element swaps the upper
and lower half of samples and can be used for example for a
wide-stereo effect. Fixes #396057
* 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:
Add docs for the audioinvert element and add them to the build system.
* tests/check/Makefile.am:
* tests/check/elements/audioinvert.c: (setup_invert),
(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
Add unit test suite for the audioinvert element.
2007-01-23 18:16:09 +00:00

278 lines
8.2 KiB
C

/*
* 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
* @short_description: Swaps upper and lower half of audio samples
*
* <refsect2>
* 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.
* <title>Example launch line</title>
* <para>
* <programlisting>
* 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
* </programlisting>
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstbasetransform.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 ("AudioInvert",
"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
};
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;
}