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/* GStreamer
* (c) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* (c) 2005 Tim-Philipp Müller <tim centricular 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-gconfaudiosrc
* @see_also: #GstAlsaSrc, #GstAutoAudioSrc
*
* This element records sound from the audiosink that has been configured in
* GConf by the user.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch gconfaudiosrc ! audioconvert ! wavenc ! filesink location=record.wav
* ]| Record from configured audioinput
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include "gstgconfelements.h"
#include "gstgconfaudiosrc.h"
static void gst_gconf_audio_src_dispose (GObject * object);
static void gst_gconf_audio_src_finalize (GstGConfAudioSrc * src);
static void cb_toggle_element (GConfClient * client,
guint connection_id, GConfEntry * entry, gpointer data);
static GstStateChangeReturn
gst_gconf_audio_src_change_state (GstElement * element,
GstStateChange transition);
GST_BOILERPLATE (GstGConfAudioSrc, gst_gconf_audio_src, GstSwitchSrc,
GST_TYPE_SWITCH_SRC);
static void
gst_gconf_audio_src_base_init (gpointer klass)
{
GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
gst_element_class_set_details_simple (eklass, "GConf audio source",
"Source/Audio",
"Audio source embedding the GConf-settings for audio input",
"GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
}
static void
gst_gconf_audio_src_class_init (GstGConfAudioSrcClass * klass)
{
GObjectClass *oklass = G_OBJECT_CLASS (klass);
GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
oklass->dispose = gst_gconf_audio_src_dispose;
oklass->finalize = (GObjectFinalizeFunc) gst_gconf_audio_src_finalize;
eklass->change_state = gst_gconf_audio_src_change_state;
}
/*
* Hack to make negotiation work.
*/
static gboolean
gst_gconf_audio_src_reset (GstGConfAudioSrc * src)
{
gst_switch_src_set_child (GST_SWITCH_SRC (src), NULL);
g_free (src->gconf_str);
src->gconf_str = NULL;
return TRUE;
}
static void
gst_gconf_audio_src_init (GstGConfAudioSrc * src,
GstGConfAudioSrcClass * g_class)
{
gst_gconf_audio_src_reset (src);
src->client = gconf_client_get_default ();
gconf_client_add_dir (src->client, GST_GCONF_DIR,
GCONF_CLIENT_PRELOAD_RECURSIVE, NULL);
src->gconf_notify_id = gconf_client_notify_add (src->client,
GST_GCONF_DIR "/" GST_GCONF_AUDIOSRC_KEY,
cb_toggle_element, src, NULL, NULL);
}
static void
gst_gconf_audio_src_dispose (GObject * object)
{
GstGConfAudioSrc *src = GST_GCONF_AUDIO_SRC (object);
if (src->client) {
if (src->gconf_notify_id) {
gconf_client_notify_remove (src->client, src->gconf_notify_id);
src->gconf_notify_id = 0;
}
g_object_unref (G_OBJECT (src->client));
src->client = NULL;
}
GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
}
static void
gst_gconf_audio_src_finalize (GstGConfAudioSrc * src)
{
g_free (src->gconf_str);
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, ((GObject *) (src)));
}
static gboolean
do_toggle_element (GstGConfAudioSrc * src)
{
GstElement *new_kid;
gchar *new_gconf_str;
new_gconf_str = gst_gconf_get_string (GST_GCONF_AUDIOSRC_KEY);
if (new_gconf_str != NULL && src->gconf_str != NULL &&
(strlen (new_gconf_str) == 0 ||
strcmp (src->gconf_str, new_gconf_str) == 0)) {
g_free (new_gconf_str);
GST_DEBUG_OBJECT (src, "GConf key was updated, but it didn't change");
return TRUE;
}
GST_DEBUG_OBJECT (src, "GConf key changed: '%s' to '%s'",
GST_STR_NULL (src->gconf_str), GST_STR_NULL (new_gconf_str));
GST_DEBUG_OBJECT (src, "Creating new kid");
if (!(new_kid = gst_gconf_get_default_audio_src ())) {
GST_ELEMENT_ERROR (src, LIBRARY, SETTINGS, (NULL),
("Failed to render audio src from GConf"));
return FALSE;
}
if (!gst_switch_src_set_child (GST_SWITCH_SRC (src), new_kid)) {
GST_WARNING_OBJECT (src, "Failed to update child element");
goto fail;
}
g_free (src->gconf_str);
src->gconf_str = new_gconf_str;
GST_DEBUG_OBJECT (src, "done changing gconf audio src");
return TRUE;
fail:
g_free (new_gconf_str);
return FALSE;
}
static void
cb_toggle_element (GConfClient * client,
guint connection_id, GConfEntry * entry, gpointer data)
{
do_toggle_element (GST_GCONF_AUDIO_SRC (data));
}
static GstStateChangeReturn
gst_gconf_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstGConfAudioSrc *src = GST_GCONF_AUDIO_SRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!do_toggle_element (src)) {
gst_gconf_audio_src_reset (src);
return GST_STATE_CHANGE_FAILURE;
}
break;
default:
break;
}
ret = GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS, change_state,
(element, transition), GST_STATE_CHANGE_SUCCESS);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
if (!gst_gconf_audio_src_reset (src))
ret = GST_STATE_CHANGE_FAILURE;
break;
default:
break;
}
return ret;
}
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