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254 lines
7.2 KiB
C
254 lines
7.2 KiB
C
/* GStreamer
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* (c) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
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* (c) 2005 Tim-Philipp Müller <tim centricular 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-gconfaudiosrc
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* @see_also: #GstAlsaSrc, #GstAutoAudioSrc
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*
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* This element records sound from the audiosink that has been configured in
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* GConf by the user.
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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 gconfaudiosrc ! audioconvert ! wavenc ! filesink location=record.wav
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* ]| Record from configured audioinput
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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 <string.h>
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#include "gstgconfelements.h"
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#include "gstgconfaudiosrc.h"
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static void gst_gconf_audio_src_dispose (GObject * object);
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static void gst_gconf_audio_src_finalize (GstGConfAudioSrc * src);
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static void cb_toggle_element (GConfClient * client,
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guint connection_id, GConfEntry * entry, gpointer data);
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static GstStateChangeReturn
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gst_gconf_audio_src_change_state (GstElement * element,
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GstStateChange transition);
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GST_BOILERPLATE (GstGConfAudioSrc, gst_gconf_audio_src, GstBin, GST_TYPE_BIN);
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static void
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gst_gconf_audio_src_base_init (gpointer klass)
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{
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GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS_ANY);
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gst_element_class_add_pad_template (eklass,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_details_simple (eklass, "GConf audio source",
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"Source/Audio",
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"Audio source embedding the GConf-settings for audio input",
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"GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
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}
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static void
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gst_gconf_audio_src_class_init (GstGConfAudioSrcClass * klass)
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{
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GObjectClass *oklass = G_OBJECT_CLASS (klass);
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GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
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oklass->dispose = gst_gconf_audio_src_dispose;
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oklass->finalize = (GObjectFinalizeFunc) gst_gconf_audio_src_finalize;
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eklass->change_state = gst_gconf_audio_src_change_state;
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}
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/*
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* Hack to make negotiation work.
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*/
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static gboolean
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gst_gconf_audio_src_reset (GstGConfAudioSrc * src)
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{
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GstPad *targetpad;
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/* fakesrc */
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if (src->kid) {
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gst_element_set_state (src->kid, GST_STATE_NULL);
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gst_bin_remove (GST_BIN (src), src->kid);
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}
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src->kid = gst_element_factory_make ("fakesrc", "testsrc");
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if (!src->kid) {
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GST_ERROR_OBJECT (src, "Failed to create fakesrc");
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return FALSE;
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}
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gst_bin_add (GST_BIN (src), src->kid);
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targetpad = gst_element_get_static_pad (src->kid, "src");
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gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
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gst_object_unref (targetpad);
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g_free (src->gconf_str);
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src->gconf_str = NULL;
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return TRUE;
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}
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static void
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gst_gconf_audio_src_init (GstGConfAudioSrc * src,
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GstGConfAudioSrcClass * g_class)
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{
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src->pad = gst_ghost_pad_new_no_target ("src", GST_PAD_SRC);
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gst_element_add_pad (GST_ELEMENT (src), src->pad);
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gst_gconf_audio_src_reset (src);
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src->client = gconf_client_get_default ();
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gconf_client_add_dir (src->client, GST_GCONF_DIR,
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GCONF_CLIENT_PRELOAD_RECURSIVE, NULL);
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src->gconf_notify_id = gconf_client_notify_add (src->client,
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GST_GCONF_DIR "/" GST_GCONF_AUDIOSRC_KEY,
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cb_toggle_element, src, NULL, NULL);
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}
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static void
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gst_gconf_audio_src_dispose (GObject * object)
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{
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GstGConfAudioSrc *src = GST_GCONF_AUDIO_SRC (object);
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if (src->client) {
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if (src->gconf_notify_id) {
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gconf_client_notify_remove (src->client, src->gconf_notify_id);
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src->gconf_notify_id = 0;
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}
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g_object_unref (G_OBJECT (src->client));
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src->client = NULL;
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}
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GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
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}
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static void
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gst_gconf_audio_src_finalize (GstGConfAudioSrc * src)
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{
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g_free (src->gconf_str);
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GST_CALL_PARENT (G_OBJECT_CLASS, finalize, ((GObject *) (src)));
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}
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static gboolean
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do_toggle_element (GstGConfAudioSrc * src)
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{
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GstState cur, next;
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GstPad *targetpad;
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gchar *new_gconf_str;
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new_gconf_str = gst_gconf_get_string (GST_GCONF_AUDIOSRC_KEY);
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if (new_gconf_str != NULL && src->gconf_str != NULL &&
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(strlen (new_gconf_str) == 0 ||
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strcmp (src->gconf_str, new_gconf_str) == 0)) {
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g_free (new_gconf_str);
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GST_DEBUG_OBJECT (src, "GConf key was updated, but it didn't change");
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return TRUE;
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}
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GST_OBJECT_LOCK (src);
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cur = GST_STATE (src);
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next = GST_STATE_PENDING (src);
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GST_OBJECT_UNLOCK (src);
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if (cur >= GST_STATE_READY || next == GST_STATE_PAUSED) {
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GST_DEBUG_OBJECT (src, "already running, ignoring GConf change");
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g_free (new_gconf_str);
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return TRUE;
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}
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GST_DEBUG_OBJECT (src, "GConf key changed: '%s' to '%s'",
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GST_STR_NULL (src->gconf_str), GST_STR_NULL (new_gconf_str));
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g_free (src->gconf_str);
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src->gconf_str = new_gconf_str;
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/* kill old element */
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if (src->kid) {
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GST_DEBUG_OBJECT (src, "Removing old kid");
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gst_element_set_state (src->kid, GST_STATE_NULL);
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gst_bin_remove (GST_BIN (src), src->kid);
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src->kid = NULL;
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}
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GST_DEBUG_OBJECT (src, "Creating new kid");
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if (!(src->kid = gst_gconf_get_default_audio_src ())) {
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GST_ELEMENT_ERROR (src, LIBRARY, SETTINGS, (NULL),
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("Failed to render audio source from GConf"));
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g_free (src->gconf_str);
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src->gconf_str = NULL;
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return FALSE;
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}
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gst_element_set_state (src->kid, GST_STATE (src));
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gst_bin_add (GST_BIN (src), src->kid);
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/* re-attach ghostpad */
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GST_DEBUG_OBJECT (src, "Creating new ghostpad");
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targetpad = gst_element_get_static_pad (src->kid, "src");
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gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
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gst_object_unref (targetpad);
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GST_DEBUG_OBJECT (src, "done changing gconf audio source");
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return TRUE;
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}
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static void
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cb_toggle_element (GConfClient * client,
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guint connection_id, GConfEntry * entry, gpointer data)
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{
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do_toggle_element (GST_GCONF_AUDIO_SRC (data));
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}
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static GstStateChangeReturn
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gst_gconf_audio_src_change_state (GstElement * element,
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GstStateChange transition)
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{
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GstGConfAudioSrc *src = GST_GCONF_AUDIO_SRC (element);
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:
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if (!do_toggle_element (src))
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return GST_STATE_CHANGE_FAILURE;
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break;
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default:
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break;
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}
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ret = GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS, change_state,
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(element, transition), GST_STATE_CHANGE_SUCCESS);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:
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if (!gst_gconf_audio_src_reset (src))
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ret = GST_STATE_CHANGE_FAILURE;
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break;
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default:
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break;
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}
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return ret;
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}
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