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webrtc: Add some locks to setters and remove non-existing functions from headers
https://bugzilla.gnome.org/show_bug.cgi?id=794363
This commit is contained in:
parent
dabfe399eb
commit
950ead9215
7 changed files with 14 additions and 15 deletions
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@ -66,7 +66,9 @@ gst_webrtc_dtls_transport_set_transport (GstWebRTCDTLSTransport * transport,
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_ICE_TRANSPORT (ice));
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g_return_if_fail (GST_IS_WEBRTC_ICE_TRANSPORT (ice));
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GST_OBJECT_LOCK (transport);
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gst_object_replace ((GstObject **) & transport->transport, GST_OBJECT (ice));
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gst_object_replace ((GstObject **) & transport->transport, GST_OBJECT (ice));
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GST_OBJECT_UNLOCK (transport);
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}
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}
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static void
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static void
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@ -66,7 +66,9 @@ void
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gst_webrtc_ice_transport_connection_state_change (GstWebRTCICETransport * ice,
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gst_webrtc_ice_transport_connection_state_change (GstWebRTCICETransport * ice,
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GstWebRTCICEConnectionState new_state)
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GstWebRTCICEConnectionState new_state)
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{
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{
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GST_OBJECT_LOCK (ice);
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ice->state = new_state;
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ice->state = new_state;
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GST_OBJECT_UNLOCK (ice);
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g_object_notify (G_OBJECT (ice), "state");
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g_object_notify (G_OBJECT (ice), "state");
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}
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}
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@ -74,7 +76,9 @@ void
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gst_webrtc_ice_transport_gathering_state_change (GstWebRTCICETransport * ice,
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gst_webrtc_ice_transport_gathering_state_change (GstWebRTCICETransport * ice,
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GstWebRTCICEGatheringState new_state)
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GstWebRTCICEGatheringState new_state)
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{
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{
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GST_OBJECT_LOCK (ice);
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ice->gathering_state = new_state;
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ice->gathering_state = new_state;
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GST_OBJECT_UNLOCK (ice);
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g_object_notify (G_OBJECT (ice), "gathering-state");
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g_object_notify (G_OBJECT (ice), "gathering-state");
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}
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}
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@ -60,8 +60,10 @@ gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
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g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
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g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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GST_OBJECT_LOCK (receiver);
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gst_object_replace ((GstObject **) & receiver->transport,
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gst_object_replace ((GstObject **) & receiver->transport,
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GST_OBJECT (transport));
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GST_OBJECT (transport));
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GST_OBJECT_UNLOCK (receiver);
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}
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}
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void
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void
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@ -71,8 +73,10 @@ gst_webrtc_rtp_receiver_set_rtcp_transport (GstWebRTCRTPReceiver * receiver,
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g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
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g_return_if_fail (GST_IS_WEBRTC_RTP_RECEIVER (receiver));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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GST_OBJECT_LOCK (receiver);
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gst_object_replace ((GstObject **) & receiver->rtcp_transport,
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gst_object_replace ((GstObject **) & receiver->rtcp_transport,
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GST_OBJECT (transport));
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GST_OBJECT (transport));
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GST_OBJECT_UNLOCK (receiver);
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}
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}
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static void
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static void
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@ -59,12 +59,6 @@ struct _GstWebRTCRTPReceiverClass
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GST_WEBRTC_API
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GST_WEBRTC_API
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GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
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GstWebRTCRTPReceiver * gst_webrtc_rtp_receiver_new (void);
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GST_WEBRTC_API
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GST_WEBRTC_API
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GstStructure * gst_webrtc_rtp_receiver_get_parameters (GstWebRTCRTPReceiver * receiver, gchar * kind);
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/* FIXME: promise? */
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GST_WEBRTC_API
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gboolean gst_webrtc_rtp_receiver_set_parameters (GstWebRTCRTPReceiver * receiver,
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GstStructure * parameters);
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GST_WEBRTC_API
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void gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
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void gst_webrtc_rtp_receiver_set_transport (GstWebRTCRTPReceiver * receiver,
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GstWebRTCDTLSTransport * transport);
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GstWebRTCDTLSTransport * transport);
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GST_WEBRTC_API
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GST_WEBRTC_API
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@ -66,8 +66,10 @@ gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
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g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
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g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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GST_OBJECT_LOCK (sender);
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gst_object_replace ((GstObject **) & sender->transport,
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gst_object_replace ((GstObject **) & sender->transport,
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GST_OBJECT (transport));
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GST_OBJECT (transport));
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GST_OBJECT_UNLOCK (sender);
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}
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}
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void
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void
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@ -77,8 +79,10 @@ gst_webrtc_rtp_sender_set_rtcp_transport (GstWebRTCRTPSender * sender,
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g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
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g_return_if_fail (GST_IS_WEBRTC_RTP_SENDER (sender));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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g_return_if_fail (GST_IS_WEBRTC_DTLS_TRANSPORT (transport));
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GST_OBJECT_LOCK (sender);
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gst_object_replace ((GstObject **) & sender->rtcp_transport,
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gst_object_replace ((GstObject **) & sender->rtcp_transport,
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GST_OBJECT (transport));
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GST_OBJECT (transport));
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GST_OBJECT_UNLOCK (sender);
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}
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}
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static void
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static void
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@ -57,12 +57,6 @@ struct _GstWebRTCRTPSenderClass
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GST_WEBRTC_API
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GST_WEBRTC_API
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GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (GArray * send_encodings);
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GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (GArray * send_encodings);
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GST_WEBRTC_API
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GstStructure * gst_webrtc_rtp_sender_get_parameters (GstWebRTCRTPSender * sender, gchar * kind);
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/* FIXME: promise? */
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GST_WEBRTC_API
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gboolean gst_webrtc_rtp_sender_set_parameters (GstWebRTCRTPSender * sender,
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GstStructure * parameters);
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GST_WEBRTC_API
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GST_WEBRTC_API
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void gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
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void gst_webrtc_rtp_sender_set_transport (GstWebRTCRTPSender * sender,
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@ -61,9 +61,6 @@ struct _GstWebRTCRTPTransceiverClass
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gpointer _padding[GST_PADDING];
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gpointer _padding[GST_PADDING];
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};
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};
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GST_WEBRTC_API
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void gst_webrtc_rtp_transceiver_stop (GstWebRTCRTPTransceiver * transceiver);
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G_END_DECLS
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G_END_DECLS
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#endif /* __GST_WEBRTC_RTP_TRANSCEIVER_H__ */
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#endif /* __GST_WEBRTC_RTP_TRANSCEIVER_H__ */
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