This will prevent any unsafe access.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2241>
#include "webrtcdatachannel.h"
#include "sctptransport.h"
+#include "gst/webrtc/webrtc-priv.h"
+
#include <gst/rtp/rtp.h>
#include <stdio.h>
#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);
#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);
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__ */