gstreamer/ext/hal/gsthalaudiosrc.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

258 lines
7.3 KiB
C

/* GStreamer
* (c) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* (c) 2005 Tim-Philipp Müller <tim centricular net>
* (c) 2006 Jürg Billeter <j@bitron.ch>
*
* 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-halaudiosrc
*
* <refsect2>
* <para>
* HalAudioSrc allows access to input of sound devices by specifying the
* corresponding persistent Unique Device Id (UDI) from the Hardware Abstraction
* Layer (HAL) in the <link linkend="GstHalAudioSrc--udi">udi</link> property.
* It currently always embeds alsasrc or osssrc as HAL doesn't support other
* sound systems yet. You can also specify the UDI of a device that has ALSA or
* OSS subdevices. If both are present ALSA is preferred.
* </para>
* <title>Examples</title>
* <para>
* To list the UDIs of all your ALSA input devices :
* <programlisting>
* hal-find-by-property --key alsa.type --string capture
* </programlisting>
* Here is a pipeline to test your sound input :
* <programlisting>
* gst-launch -v halaudiosrc udi=/org/freedesktop/Hal/devices/pci_8086_27d8_alsa_capture_0 ! autoaudiosink
* </programlisting>
* You should now hear yourself with a small delay if you have a microphone
* connected to the specified sound device.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gsthalelements.h"
#include "gsthalaudiosrc.h"
static void gst_hal_audio_src_dispose (GObject * object);
static GstStateChangeReturn
gst_hal_audio_src_change_state (GstElement * element,
GstStateChange transition);
enum
{
PROP_0,
PROP_UDI
};
GST_BOILERPLATE (GstHalAudioSrc, gst_hal_audio_src, GstBin, GST_TYPE_BIN);
static void gst_hal_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_hal_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void
gst_hal_audio_src_base_init (gpointer klass)
{
GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
static const GstElementDetails gst_hal_audio_src_details =
GST_ELEMENT_DETAILS ("HAL audio source",
"Source/Audio",
"Audio source for sound device access via HAL",
"Jürg Billeter <j@bitron.ch>");
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_hal_audio_src_details);
}
static void
gst_hal_audio_src_class_init (GstHalAudioSrcClass * klass)
{
GObjectClass *oklass = G_OBJECT_CLASS (klass);
GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
oklass->set_property = gst_hal_audio_src_set_property;
oklass->get_property = gst_hal_audio_src_get_property;
oklass->dispose = gst_hal_audio_src_dispose;
eklass->change_state = gst_hal_audio_src_change_state;
g_object_class_install_property (oklass, PROP_UDI,
g_param_spec_string ("udi",
"UDI", "Unique Device Id", NULL, G_PARAM_READWRITE));
}
/*
* Hack to make negotiation work.
*/
static void
gst_hal_audio_src_reset (GstHalAudioSrc * 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");
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);
}
static void
gst_hal_audio_src_init (GstHalAudioSrc * src, GstHalAudioSrcClass * g_class)
{
src->pad = gst_ghost_pad_new_no_target ("src", GST_PAD_SRC);
gst_element_add_pad (GST_ELEMENT (src), src->pad);
gst_hal_audio_src_reset (src);
}
static void
gst_hal_audio_src_dispose (GObject * object)
{
GstHalAudioSrc *src = GST_HAL_AUDIO_SRC (object);
if (src->udi) {
g_free (src->udi);
src->udi = NULL;
}
GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
}
static gboolean
do_toggle_element (GstHalAudioSrc * src)
{
GstPad *targetpad;
/* 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->udi)
GST_INFO_OBJECT (src, "No UDI set for device, using default one");
if (!(src->kid = gst_hal_get_audio_src (src->udi))) {
GST_ELEMENT_ERROR (src, LIBRARY, SETTINGS, (NULL),
("Failed to render audio source from Hal"));
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 hal audio source");
return TRUE;
}
static void
gst_hal_audio_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstHalAudioSrc *this = GST_HAL_AUDIO_SRC (object);
GST_OBJECT_LOCK (this);
switch (prop_id) {
case PROP_UDI:
if (this->udi)
g_free (this->udi);
this->udi = g_value_dup_string (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (this);
}
static void
gst_hal_audio_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstHalAudioSrc *this = GST_HAL_AUDIO_SRC (object);
GST_OBJECT_LOCK (this);
switch (prop_id) {
case PROP_UDI:
g_value_set_string (value, this->udi);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (this);
}
static GstStateChangeReturn
gst_hal_audio_src_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstHalAudioSrc *src = GST_HAL_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:
gst_hal_audio_src_reset (src);
break;
default:
break;
}
return ret;
}