mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-02-02 20:42:30 +00:00
Code Issues 1 Projects Releases Wiki Activity

webrtc: Remove APIs to set transport on sender/receiver

Browse source 
They're not not used ever.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1765>
This commit is contained in:
Olivier Crête 2020-11-02 19:55:46 -05:00 committed by GStreamer Merge Bot
parent 5d5417f271
commit c5d76d944e
4 changed files with 0 additions and 36 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

15
gst-libs/gst/webrtc/rtpreceiver.c
View file

@ -53,21 +53,6 @@ enum
//static guint gst_webrtc_rtp_receiver_signals[LAST_SIGNAL] = { 0 };
void
gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
GST_OBJECT_LOCK (receiver);
gst_object_replace ((GstObject **) & receiver->transport,
GST_OBJECT (transport));
gst_object_replace ((GstObject **) & receiver->rtcp_transport,
GST_OBJECT (transport));
GST_OBJECT_UNLOCK (receiver);
}
static void
gst_webrtc_rtp_receiver_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)

3
gst-libs/gst/webrtc/rtpreceiver.h
View file

@ -66,9 +66,6 @@ struct _GstWebRTCRTPReceiverClass
GST_WEBRTC_API
GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
GST_WEBRTC_API
void gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
GstWebRTCDTLSTransport * transport);
G_DEFINE_AUTOPTR_CLEANUP_FUNC(GstWebRTCRTPReceiver, gst_object_unref)

15
gst-libs/gst/webrtc/rtpsender.c
View file

@ -56,21 +56,6 @@ enum
//static guint gst_webrtc_rtp_sender_signals[LAST_SIGNAL] = { 0 };
void
gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport)
{
g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
GST_OBJECT_LOCK (sender);
gst_object_replace ((GstObject **) & sender->transport,
GST_OBJECT (transport));
gst_object_replace ((GstObject **) & sender->rtcp_transport,
GST_OBJECT (transport));
GST_OBJECT_UNLOCK (sender);
}
/**
* gst_webrtc_rtp_sender_set_priority:
* @sender: a #GstWebRTCRTPSender

3
gst-libs/gst/webrtc/rtpsender.h
View file

@ -79,9 +79,6 @@ struct _GstWebRTCRTPSenderClass
GST_WEBRTC_API
GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void);
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
GstWebRTCDTLSTransport * transport);
GST_WEBRTC_API
void gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender *sender,
GstWebRTCPriorityType priority);