gstreamer/ext/gconf/gstgconfaudiosrc.c
Wim Taymans 487b784b4f Don't use gst_element_get_pad(), it's a bad method.
Original commit message from CVS:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
(do_toggle_element):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
(do_toggle_element):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
(do_toggle_element):
* ext/gconf/gstswitchsink.c: (gst_switch_commit_new_kid):
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_reset),
(do_toggle_element):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_reset),
(do_toggle_element):
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_stream_free), (gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink), (gst_rtspsrc_skip_lws),
(gst_rtspsrc_unskip_lws), (gst_rtspsrc_skip_commas),
(gst_rtspsrc_skip_item), (gst_rtsp_decode_quoted_string),
(gst_rtspsrc_parse_digest_challenge), (gst_rtspsrc_parse_auth_hdr):
* tests/icles/videocrop-test.c: (test_with_caps),
(video_crop_get_test_caps):
Don't use gst_element_get_pad(), it's a bad method.
2008-05-21 17:39:38 +00:00

241 lines
7 KiB
C

/* 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.
*/
#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, GstBin, GST_TYPE_BIN);
static void
gst_gconf_audio_src_base_init (gpointer klass)
{
GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
static const GstElementDetails gst_gconf_audio_src_details =
GST_ELEMENT_DETAILS ("GConf audio source",
"Source/Audio",
"Audio source embedding the GConf-settings for audio input",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
gst_element_class_add_pad_template (eklass,
gst_static_pad_template_get (&src_template));
gst_element_class_set_details (eklass, &gst_gconf_audio_src_details);
}
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)
{
GstPad *targetpad;
/* fakesrc */
if (src->kid) {
gst_element_set_state (src->kid, GST_STATE_NULL);
gst_bin_remove (GST_BIN (src), src->kid);
}
src->kid = gst_element_factory_make ("fakesrc", "testsrc");
if (!src->kid) {
GST_ERROR_OBJECT (src, "Failed to create fakesrc");
return FALSE;
}
gst_bin_add (GST_BIN (src), src->kid);
targetpad = gst_element_get_static_pad (src->kid, "src");
gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
gst_object_unref (targetpad);
g_free (src->gconf_str);
src->gconf_str = NULL;
return TRUE;
}
static void
gst_gconf_audio_src_init (GstGConfAudioSrc * src,
GstGConfAudioSrcClass * g_class)
{
src->pad = gst_ghost_pad_new_no_target ("src", GST_PAD_SRC);
gst_element_add_pad (GST_ELEMENT (src), src->pad);
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)
{
GstState cur, next;
GstPad *targetpad;
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_OBJECT_LOCK (src);
cur = GST_STATE (src);
next = GST_STATE_PENDING (src);
GST_OBJECT_UNLOCK (src);
if (cur >= GST_STATE_READY || next == GST_STATE_PAUSED) {
GST_DEBUG_OBJECT (src, "already running, ignoring GConf change");
g_free (new_gconf_str);
return TRUE;
}
GST_DEBUG_OBJECT (src, "GConf key changed: '%s' to '%s'",
GST_STR_NULL (src->gconf_str), GST_STR_NULL (new_gconf_str));
g_free (src->gconf_str);
src->gconf_str = new_gconf_str;
/* kill old element */
if (src->kid) {
GST_DEBUG_OBJECT (src, "Removing old kid");
gst_element_set_state (src->kid, GST_STATE_NULL);
gst_bin_remove (GST_BIN (src), src->kid);
src->kid = NULL;
}
GST_DEBUG_OBJECT (src, "Creating new kid");
if (!(src->kid = gst_gconf_get_default_audio_src ())) {
GST_ELEMENT_ERROR (src, LIBRARY, SETTINGS, (NULL),
("Failed to render audio source from GConf"));
g_free (src->gconf_str);
src->gconf_str = NULL;
return FALSE;
}
gst_element_set_state (src->kid, GST_STATE (src));
gst_bin_add (GST_BIN (src), src->kid);
/* re-attach ghostpad */
GST_DEBUG_OBJECT (src, "Creating new ghostpad");
targetpad = gst_element_get_static_pad (src->kid, "src");
gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
gst_object_unref (targetpad);
GST_DEBUG_OBJECT (src, "done changing gconf audio source");
return TRUE;
}
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))
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;
}