diff options
author | Olivier CrĂȘte <olivier.crete@collabora.com> | 2021-04-21 16:00:57 -0400 |
---|---|---|
committer | GStreamer Marge Bot <gitlab-merge-bot@gstreamer-foundation.org> | 2021-06-21 20:53:09 +0000 |
commit | a6593753a5ab751eb10971201960f3e8aea70bcd (patch) | |
tree | f71c36ae536072c68c97587adacc2178c1d0f8cf /gst-libs/gst | |
parent | b5f2de31244e86cb02a8c790a2310f9138965665 (diff) | |
download | gstreamer-plugins-bad-a6593753a5ab751eb10971201960f3e8aea70bcd.tar.gz |
webrtc lib: Make the rtpsender struct private
This will prevent any unsafe access.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
Diffstat (limited to 'gst-libs/gst')
-rw-r--r-- | gst-libs/gst/webrtc/rtpsender.c | 1 | ||||
-rw-r--r-- | gst-libs/gst/webrtc/rtpsender.h | 42 | ||||
-rw-r--r-- | gst-libs/gst/webrtc/webrtc-priv.h | 42 |
3 files changed, 43 insertions, 42 deletions
diff --git a/gst-libs/gst/webrtc/rtpsender.c b/gst-libs/gst/webrtc/rtpsender.c index 90221dd56..90f43ccee 100644 --- a/gst-libs/gst/webrtc/rtpsender.c +++ b/gst-libs/gst/webrtc/rtpsender.c @@ -32,6 +32,7 @@ #include "rtpsender.h" #include "rtptransceiver.h" +#include "webrtc-priv.h" #define GST_CAT_DEFAULT gst_webrtc_rtp_sender_debug GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); diff --git a/gst-libs/gst/webrtc/rtpsender.h b/gst-libs/gst/webrtc/rtpsender.h index 44eee81a5..b3ca9a010 100644 --- a/gst-libs/gst/webrtc/rtpsender.h +++ b/gst-libs/gst/webrtc/rtpsender.h @@ -35,48 +35,6 @@ GType gst_webrtc_rtp_sender_get_type(void); #define GST_IS_WEBRTC_RTP_SENDER_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_WEBRTC_RTP_SENDER)) #define GST_WEBRTC_RTP_SENDER_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_WEBRTC_RTP_SENDER,GstWebRTCRTPSenderClass)) -/** - * GstWebRTCRTPSender: - * @transport: The transport for RTP packets - * @send_encodings: Unused - * @priority: The priority of the stream (Since: 1.20) - * - * An object to track the sending aspect of the stream - * - * Mostly matches the WebRTC RTCRtpSender interface. - * - * Since: 1.16 - */ -/** - * GstWebRTCRTPSender.priority: - * - * The priority of the stream - * - * Since: 1.20 - */ -struct _GstWebRTCRTPSender -{ - GstObject parent; - - /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */ - GstWebRTCDTLSTransport *transport; - - GArray *send_encodings; - GstWebRTCPriorityType priority; - - gpointer _padding[GST_PADDING]; -}; - -struct _GstWebRTCRTPSenderClass -{ - GstObjectClass parent_class; - - gpointer _padding[GST_PADDING]; -}; - -GST_WEBRTC_API -GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void); - GST_WEBRTC_API void gst_webrtc_rtp_sender_set_priority (GstWebRTCRTPSender *sender, GstWebRTCPriorityType priority); diff --git a/gst-libs/gst/webrtc/webrtc-priv.h b/gst-libs/gst/webrtc/webrtc-priv.h index 6f2d3ea8d..559fc2d3a 100644 --- a/gst-libs/gst/webrtc/webrtc-priv.h +++ b/gst-libs/gst/webrtc/webrtc-priv.h @@ -86,6 +86,48 @@ struct _GstWebRTCRTPTransceiverClass gpointer _padding[GST_PADDING]; }; +/** + * GstWebRTCRTPSender: + * @transport: The transport for RTP packets + * @send_encodings: Unused + * @priority: The priority of the stream (Since: 1.20) + * + * An object to track the sending aspect of the stream + * + * Mostly matches the WebRTC RTCRtpSender interface. + * + * Since: 1.16 + */ +/** + * GstWebRTCRTPSender.priority: + * + * The priority of the stream + * + * Since: 1.20 + */ +struct _GstWebRTCRTPSender +{ + GstObject parent; + + /* The MediStreamTrack is represented by the stream and is output into @transport as necessary */ + GstWebRTCDTLSTransport *transport; + + GArray *send_encodings; + GstWebRTCPriorityType priority; + + gpointer _padding[GST_PADDING]; +}; + +struct _GstWebRTCRTPSenderClass +{ + GstObjectClass parent_class; + + gpointer _padding[GST_PADDING]; +}; + +GST_WEBRTC_API +GstWebRTCRTPSender * gst_webrtc_rtp_sender_new (void); + G_END_DECLS #endif /* __GST_WEBRTC_PRIV_H__ */ |