Skip to content
Permalink
Branch: master
Find file Copy path
Find file Copy path
Fetching contributors…
Cannot retrieve contributors at this time
257 lines (213 sloc) 7.56 KB
/*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-audioinvert
* @title: 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-1.0 audiotestsrc wave=saw ! audioinvert degree=0.4 ! alsasink
* gst-launch-1.0 filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert degree=0.4 ! alsasink
* gst-launch-1.0 audiotestsrc wave=saw ! audioconvert ! audioinvert degree=0.4 ! audioconvert ! alsasink
* ]|
*
*/
#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 "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," \
" format=(string) {"GST_AUDIO_NE(S16)","GST_AUDIO_NE(F32)"}," \
" rate=(int)[1,MAX]," \
" channels=(int)[1,MAX]," \
" layout=(string) {interleaved, non-interleaved}"
G_DEFINE_TYPE (GstAudioInvert, gst_audio_invert, GST_TYPE_AUDIO_FILTER);
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,
const GstAudioInfo * info);
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_class_init (GstAudioInvertClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstCaps *caps;
GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0,
"audioinvert element");
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) 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_element_class_set_static_metadata (gstelement_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);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
GST_DEBUG_FUNCPTR (gst_audio_invert_transform_ip);
GST_BASE_TRANSFORM_CLASS (klass)->transform_ip_on_passthrough = FALSE;
GST_AUDIO_FILTER_CLASS (klass)->setup =
GST_DEBUG_FUNCPTR (gst_audio_invert_setup);
}
static void
gst_audio_invert_init (GstAudioInvert * filter)
{
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, const GstAudioInfo * info)
{
GstAudioInvert *filter = GST_AUDIO_INVERT (base);
gboolean ret = TRUE;
switch (GST_AUDIO_INFO_FORMAT (info)) {
case GST_AUDIO_FORMAT_S16:
filter->process = (GstAudioInvertProcessFunc)
gst_audio_invert_transform_int;
break;
case GST_AUDIO_FORMAT_F32:
filter->process = (GstAudioInvertProcessFunc)
gst_audio_invert_transform_float;
break;
default:
ret = FALSE;
break;
}
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;
GstMapInfo map;
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 (GST_OBJECT (filter), stream_time);
if (G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
return GST_FLOW_OK;
gst_buffer_map (buf, &map, GST_MAP_READWRITE);
num_samples = map.size / GST_AUDIO_FILTER_BPS (filter);
filter->process (filter, map.data, num_samples);
gst_buffer_unmap (buf, &map);
return GST_FLOW_OK;
}
You can’t perform that action at this time.