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rtpsender: Add API to set the priority

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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1707>
This commit is contained in:
Olivier Crête 2020-07-09 13:39:03 -04:00
parent 0fbbdc5734
commit cca313ecd8
2 changed files with 68 additions and 12 deletions
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68
gst-libs/gst/webrtc/rtpsender.c
View file

@ -51,10 +51,7 @@ enum
enum
{
PROP_0,
PROP_MID,
PROP_SENDER,
PROP_STOPPED,
PROP_DIRECTION,
PROP_PRIORITY
};
//static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
@ -85,11 +82,38 @@ gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
GST_OBJECT_UNLOCK (sender);
}
/**
* gst_webrtc_rtp_sender_set_priority:
* @sender: a #GstWebRTCRTPSender
* @priority: The priority of this sender
*
* Sets the content of the IPv4 Type of Service (ToS), also known as DSCP
* (Differentiated Services Code Point).
* This also sets the Traffic Class field of IPv6.
*
* Since: 1.20
*/
void
gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender * sender,
GstWebRTCPriorityType priority)
{
GST_OBJECT_LOCK (sender);
sender->priority = priority;
GST_OBJECT_UNLOCK (sender);
g_object_notify (G_OBJECT (sender), "priority");
}
static void
gst_webrtc_rtp_sender_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
switch (prop_id) {
case PROP_PRIORITY:
gst_webrtc_rtp_sender_set_priority (sender, g_value_get_uint (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -100,7 +124,14 @@ static void
gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
switch (prop_id) {
case PROP_PRIORITY:
GST_OBJECT_LOCK (sender);
g_value_set_uint (value, sender->priority);
GST_OBJECT_UNLOCK (sender);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
@ -110,15 +141,15 @@ gst_webrtc_rtp_sender_get_property (GObject * object, guint prop_id,
static void
gst_webrtc_rtp_sender_finalize (GObject * object)
{
GstWebRTCRTPSender *webrtc = GST_WEBRTC_RTP_SENDER (object);
GstWebRTCRTPSender *sender = GST_WEBRTC_RTP_SENDER (object);
if (webrtc->transport)
gst_object_unref (webrtc->transport);
webrtc->transport = NULL;
if (sender->transport)
gst_object_unref (sender->transport);
sender->transport = NULL;
if (webrtc->rtcp_transport)
gst_object_unref (webrtc->rtcp_transport);
webrtc->rtcp_transport = NULL;
if (sender->rtcp_transport)
gst_object_unref (sender->rtcp_transport);
sender->rtcp_transport = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
@ -131,6 +162,21 @@ gst_webrtc_rtp_sender_class_init (GstWebRTCRTPSenderClass * klass)
gobject_class->get_property = gst_webrtc_rtp_sender_get_property;
gobject_class->set_property = gst_webrtc_rtp_sender_set_property;
gobject_class->finalize = gst_webrtc_rtp_sender_finalize;
/**
* GstWebRTCRTPSender:priority:
*
* The priority from which to set the DSCP field on packets
*
* Since: 1.20
*/
g_object_class_install_property (gobject_class,
PROP_PRIORITY,
g_param_spec_enum ("priority",
"Priority",
"The priority from which to set the DSCP field on packets",
GST_TYPE_WEBRTC_PRIORITY_TYPE, GST_WEBRTC_PRIORITY_TYPE_LOW,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static void

12
gst-libs/gst/webrtc/rtpsender.h
View file

@ -48,6 +48,13 @@ GType gst_webrtc_rtp_sender_get_type(void);
*
* Since: 1.16
*/
/**
* GstWebRTCRTPSender.priority:
*
* The priority of the stream
*
* Since: 1.20
*/
struct _GstWebRTCRTPSender
{
GstObject parent;
@ -57,6 +64,7 @@ struct _GstWebRTCRTPSender
GstWebRTCDTLSTransport *rtcp_transport;
GArray *send_encodings;
GstWebRTCPriorityType priority;
gpointer _padding[GST_PADDING];
};
@ -77,7 +85,9 @@ void gst_webrtc_rtp_sender_set_transport (GstWebR
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport);
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender *sender,
GstWebRTCPriorityType priority);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPSender, gst_object_unref)