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259 lines
8 KiB
C
259 lines
8 KiB
C
/*
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* GStreamer
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* Copyright (C) 2007 Sebastian Dröge <slomo@circular-chaos.org>
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* Copyright (C) 2006 Stefan Kost <ensonic@users.sf.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-audioinvert
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*
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* Swaps upper and lower half of audio samples. Mixing an inverted sample on top of
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* the original with a slight delay can produce effects that sound like resonance.
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* Creating a stereo sample from a mono source, with one channel inverted produces wide-stereo sounds.
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*
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* <refsect2>
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* <title>Example launch line</title>
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* |[
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* gst-launch audiotestsrc wave=saw ! audioinvert invert=0.4 ! alsasink
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* gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audioinvert invert=0.4 ! alsasink
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* gst-launch audiotestsrc wave=saw ! audioconvert ! audioinvert invert=0.4 ! audioconvert ! alsasink
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* ]|
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiofilter.h>
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#include <gst/controller/gstcontroller.h>
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#include "audioinvert.h"
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#define GST_CAT_DEFAULT gst_audio_invert_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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/* Filter signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0,
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PROP_DEGREE
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};
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#define ALLOWED_CAPS \
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"audio/x-raw-int," \
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" depth=(int)16," \
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" width=(int)16," \
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" endianness=(int)BYTE_ORDER," \
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" signed=(bool)TRUE," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]; " \
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"audio/x-raw-float," \
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" width=(int)32," \
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" endianness=(int)BYTE_ORDER," \
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" rate=(int)[1,MAX]," \
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" channels=(int)[1,MAX]"
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#define DEBUG_INIT(bla) \
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GST_DEBUG_CATEGORY_INIT (gst_audio_invert_debug, "audioinvert", 0, "audioinvert element");
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GST_BOILERPLATE_FULL (GstAudioInvert, gst_audio_invert, GstAudioFilter,
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GST_TYPE_AUDIO_FILTER, DEBUG_INIT);
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static void gst_audio_invert_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_audio_invert_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_audio_invert_setup (GstAudioFilter * filter,
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GstRingBufferSpec * format);
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static GstFlowReturn gst_audio_invert_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static void gst_audio_invert_transform_int (GstAudioInvert * filter,
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gint16 * data, guint num_samples);
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static void gst_audio_invert_transform_float (GstAudioInvert * filter,
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gfloat * data, guint num_samples);
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/* GObject vmethod implementations */
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static void
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gst_audio_invert_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstCaps *caps;
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gst_element_class_set_details_simple (element_class, "Audio inversion",
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"Filter/Effect/Audio",
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"Swaps upper and lower half of audio samples",
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"Sebastian Dröge <slomo@circular-chaos.org>");
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caps = gst_caps_from_string (ALLOWED_CAPS);
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gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass),
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caps);
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gst_caps_unref (caps);
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}
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static void
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gst_audio_invert_class_init (GstAudioInvertClass * klass)
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{
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GObjectClass *gobject_class;
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gobject_class = (GObjectClass *) klass;
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gobject_class->set_property = gst_audio_invert_set_property;
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gobject_class->get_property = gst_audio_invert_get_property;
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g_object_class_install_property (gobject_class, PROP_DEGREE,
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g_param_spec_float ("degree", "Degree",
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"Degree of inversion", 0.0, 1.0,
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0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE));
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GST_AUDIO_FILTER_CLASS (klass)->setup =
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GST_DEBUG_FUNCPTR (gst_audio_invert_setup);
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GST_BASE_TRANSFORM_CLASS (klass)->transform_ip =
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GST_DEBUG_FUNCPTR (gst_audio_invert_transform_ip);
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}
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static void
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gst_audio_invert_init (GstAudioInvert * filter, GstAudioInvertClass * klass)
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{
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filter->degree = 0.0;
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gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE);
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gst_base_transform_set_gap_aware (GST_BASE_TRANSFORM (filter), TRUE);
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}
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static void
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gst_audio_invert_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstAudioInvert *filter = GST_AUDIO_INVERT (object);
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switch (prop_id) {
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case PROP_DEGREE:
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filter->degree = g_value_get_float (value);
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gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
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filter->degree == 0.0);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_audio_invert_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstAudioInvert *filter = GST_AUDIO_INVERT (object);
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switch (prop_id) {
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case PROP_DEGREE:
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g_value_set_float (value, filter->degree);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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/* GstAudioFilter vmethod implementations */
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static gboolean
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gst_audio_invert_setup (GstAudioFilter * base, GstRingBufferSpec * format)
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{
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GstAudioInvert *filter = GST_AUDIO_INVERT (base);
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gboolean ret = TRUE;
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if (format->type == GST_BUFTYPE_FLOAT && format->width == 32)
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filter->process = (GstAudioInvertProcessFunc)
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gst_audio_invert_transform_float;
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else if (format->type == GST_BUFTYPE_LINEAR && format->width == 16)
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filter->process = (GstAudioInvertProcessFunc)
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gst_audio_invert_transform_int;
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else
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ret = FALSE;
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return ret;
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}
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static void
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gst_audio_invert_transform_int (GstAudioInvert * filter,
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gint16 * data, guint num_samples)
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{
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gint i;
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gfloat dry = 1.0 - filter->degree;
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glong val;
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for (i = 0; i < num_samples; i++) {
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val = (*data) * dry + (-1 - (*data)) * filter->degree;
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*data++ = (gint16) CLAMP (val, G_MININT16, G_MAXINT16);
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}
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}
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static void
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gst_audio_invert_transform_float (GstAudioInvert * filter,
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gfloat * data, guint num_samples)
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{
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gint i;
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gfloat dry = 1.0 - filter->degree;
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glong val;
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for (i = 0; i < num_samples; i++) {
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val = (*data) * dry - (*data) * filter->degree;
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*data++ = val;
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}
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}
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/* GstBaseTransform vmethod implementations */
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static GstFlowReturn
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gst_audio_invert_transform_ip (GstBaseTransform * base, GstBuffer * buf)
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{
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GstAudioInvert *filter = GST_AUDIO_INVERT (base);
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guint num_samples;
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GstClockTime timestamp, stream_time;
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timestamp = GST_BUFFER_TIMESTAMP (buf);
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stream_time =
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gst_segment_to_stream_time (&base->segment, GST_FORMAT_TIME, timestamp);
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GST_DEBUG_OBJECT (filter, "sync to %" GST_TIME_FORMAT,
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GST_TIME_ARGS (timestamp));
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if (GST_CLOCK_TIME_IS_VALID (stream_time))
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gst_object_sync_values (G_OBJECT (filter), stream_time);
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num_samples =
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GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8);
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if (gst_base_transform_is_passthrough (base) ||
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G_UNLIKELY (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)))
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return GST_FLOW_OK;
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filter->process (filter, GST_BUFFER_DATA (buf), num_samples);
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return GST_FLOW_OK;
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}
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