From b3eeaf5f5a2b97d96b20d3ac64b5c8c7d0693c99 Mon Sep 17 00:00:00 2001 From: =?utf8?q?Olivier=20Cr=C3=AAte?= Date: Wed, 4 Nov 2020 17:06:02 -0500 Subject: [PATCH] webrtc: Also remove rtcp_transport from the structure Change-Id: Id63ee4d28b7ebaf72004af554d16f711d6d72ee9 Part-of: Signed-off-by: Sangchul Lee --- gst-libs/gst/webrtc/rtpreceiver.c | 4 ---- gst-libs/gst/webrtc/rtpreceiver.h | 4 +--- gst-libs/gst/webrtc/rtpsender.c | 4 ---- gst-libs/gst/webrtc/rtpsender.h | 4 +--- 4 files changed, 2 insertions(+), 14 deletions(-) diff --git a/gst-libs/gst/webrtc/rtpreceiver.c b/gst-libs/gst/webrtc/rtpreceiver.c index d20239a..fb8f6e6 100644 --- a/gst-libs/gst/webrtc/rtpreceiver.c +++ b/gst-libs/gst/webrtc/rtpreceiver.c @@ -84,10 +84,6 @@ gst_webrtc_rtp_receiver_finalize (GObject * object) gst_object_unref (webrtc->transport); webrtc->transport = NULL; - if (webrtc->rtcp_transport) - gst_object_unref (webrtc->rtcp_transport); - webrtc->rtcp_transport = NULL; - G_OBJECT_CLASS (parent_class)->finalize (object); } diff --git a/gst-libs/gst/webrtc/rtpreceiver.h b/gst-libs/gst/webrtc/rtpreceiver.h index c2fa210..a02bad1 100644 --- a/gst-libs/gst/webrtc/rtpreceiver.h +++ b/gst-libs/gst/webrtc/rtpreceiver.h @@ -38,7 +38,6 @@ GType gst_webrtc_rtp_receiver_get_type(void); /** * GstWebRTCRTPReceiver: * @transport: The transport for RTP packets - * @rtcp_transport: The transport for RTCP packets without rtcp-mux * * An object to track the receiving aspect of the stream * @@ -50,9 +49,8 @@ struct _GstWebRTCRTPReceiver { GstObject parent; - /* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */ + /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */ GstWebRTCDTLSTransport *transport; - GstWebRTCDTLSTransport *rtcp_transport; gpointer _padding[GST_PADDING]; }; diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c index 40d871c..388c48d 100644 --- a/gst-libs/gst/webrtc/rtpsender.c +++ b/gst-libs/gst/webrtc/rtpsender.c @@ -121,10 +121,6 @@ gst_webrtc_rtp_sender_finalize (GObject * object) gst_object_unref (sender->transport); sender->transport = NULL; - if (sender->rtcp_transport) - gst_object_unref (sender->rtcp_transport); - sender->rtcp_transport = NULL; - G_OBJECT_CLASS (parent_class)->finalize (object); } diff --git a/gst-libs/gst/webrtc/rtpsender.h b/gst-libs/gst/webrtc/rtpsender.h index 5fa9fe8..44eee81 100644 --- a/gst-libs/gst/webrtc/rtpsender.h +++ b/gst-libs/gst/webrtc/rtpsender.h @@ -38,7 +38,6 @@ GType gst_webrtc_rtp_sender_get_type(void); /** * GstWebRTCRTPSender: * @transport: The transport for RTP packets - * @rtcp_transport: The transport for RTCP packets without rtcp-mux * @send_encodings: Unused * @priority: The priority of the stream (Since: 1.20) * @@ -59,9 +58,8 @@ struct _GstWebRTCRTPSender { GstObject parent; - /* The MediStreamTrack is represented by the stream and is output into @transport/@rtcp_transport as necessary */ + /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */ GstWebRTCDTLSTransport *transport; - GstWebRTCDTLSTransport *rtcp_transport; GArray *send_encodings; GstWebRTCPriorityType priority; -- 2.7.4