From: Olivier CrĂȘte Date: Tue, 3 Nov 2020 00:55:46 +0000 (-0500) Subject: webrtc: Remove APIs to set transport on sender/receiver X-Git-Tag: submit/tizen/20210818.024548~2 X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=7e4341b7da6761498c8ffce3a5351b23c3b96573;p=platform%2Fupstream%2Fgst-plugins-bad.git webrtc: Remove APIs to set transport on sender/receiver They're not not used ever. Change-Id: Ia90b7edfd32571bc018cb0cb2e5c1a8132da79e7 Part-of: Signed-off-by: Sangchul Lee --- diff --git a/gst-libs/gst/webrtc/rtpreceiver.c b/gst-libs/gst/webrtc/rtpreceiver.c index dd8c5a989..d20239a20 100644 --- a/gst-libs/gst/webrtc/rtpreceiver.c +++ b/gst-libs/gst/webrtc/rtpreceiver.c @@ -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) diff --git a/gst-libs/gst/webrtc/rtpreceiver.h b/gst-libs/gst/webrtc/rtpreceiver.h index c4260f73f..c2fa210f7 100644 --- a/gst-libs/gst/webrtc/rtpreceiver.h +++ b/gst-libs/gst/webrtc/rtpreceiver.h @@ -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) diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c index 3c533bf2d..40d871c32 100644 --- a/gst-libs/gst/webrtc/rtpsender.c +++ b/gst-libs/gst/webrtc/rtpsender.c @@ -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 diff --git a/gst-libs/gst/webrtc/rtpsender.h b/gst-libs/gst/webrtc/rtpsender.h index 6c42f2dac..5fa9fe83c 100644 --- a/gst-libs/gst/webrtc/rtpsender.h +++ b/gst-libs/gst/webrtc/rtpsender.h @@ -79,12 +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_rtcp_transport (GstWebRTCRTPSender * sender, - GstWebRTCDTLSTransport * transport); GST_WEBRTC_API void gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender *sender, GstWebRTCPriorityType priority);