rtsp stream transport
The GstRTSPStreamTransport configures the transport used by a GstRTSPStream. It is usually manages by a GstRTSPSessionMedia object.
With gst_rtsp_stream_transport_set_callbacks, callbacks can be configured to handle the RTP and RTCP packets from the stream, for example when they need to be sent over TCP.
With gst_rtsp_stream_transport_set_active the transports are added and removed from the stream.
A GstRTSPStream will call gst_rtsp_stream_transport_keep_alive when RTCP is received from the client. It will also call gst_rtsp_stream_transport_set_timed_out when a receiver has timed out.
A GstRTSPClient will call gst_rtsp_stream_transport_message_sent when it has sent a data message for the transport.
Last reviewed on 2013-07-16 (1.0.0)
GstRTSPStreamTransport
GObject ╰──GstRTSPStreamTransport
A Transport description for a stream
Members
parent
(GObject)
–
parent instance
Class structure
GstRTSPStreamTransportClass
Fields
parent_class
(GObjectClass)
–
GstRtspServer.RTSPStreamTransportClass
Attributes
parent_class
(GObject.ObjectClass)
–
GstRtspServer.RTSPStreamTransportClass
Attributes
parent_class
(GObject.ObjectClass)
–
GstRtspServer.RTSPStreamTransport
GObject.Object ╰──GstRtspServer.RTSPStreamTransport
A Transport description for a stream
Members
parent
(GObject.Object)
–
parent instance
GstRtspServer.RTSPStreamTransport
GObject.Object ╰──GstRtspServer.RTSPStreamTransport
A Transport description for a stream
Members
parent
(GObject.Object)
–
parent instance
Constructors
gst_rtsp_stream_transport_new
GstRTSPStreamTransport * gst_rtsp_stream_transport_new (GstRTSPStream * stream, GstRTSPTransport * tr)
Create a new GstRTSPStreamTransport that can be used to manage stream with transport tr.
a new GstRTSPStreamTransport
GstRtspServer.RTSPStreamTransport.prototype.new
function GstRtspServer.RTSPStreamTransport.prototype.new(stream: GstRtspServer.RTSPStream, tr: GstRtsp.RTSPTransport): {
// javascript wrapper for 'gst_rtsp_stream_transport_new'
}
Create a new GstRtspServer.RTSPStreamTransport that can be used to manage stream with transport tr.
Parameters:
a GstRTSPTransport
GstRtspServer.RTSPStreamTransport.new
def GstRtspServer.RTSPStreamTransport.new (stream, tr):
#python wrapper for 'gst_rtsp_stream_transport_new'
Create a new GstRtspServer.RTSPStreamTransport that can be used to manage stream with transport tr.
Parameters:
a GstRTSPTransport
Methods
gst_rtsp_stream_transport_get_rtpinfo
gchar * gst_rtsp_stream_transport_get_rtpinfo (GstRTSPStreamTransport * trans, GstClockTime start_time)
Get the RTP-Info string for trans and start_time.
GstRtspServer.RTSPStreamTransport.prototype.get_rtpinfo
function GstRtspServer.RTSPStreamTransport.prototype.get_rtpinfo(start_time: Number): {
// javascript wrapper for 'gst_rtsp_stream_transport_get_rtpinfo'
}
Get the RTP-Info string for trans and start_time.
Parameters:
a star time
the RTPInfo string for trans and start_time or null when the RTP-Info could not be determined. GLib.prototype.free after usage.
GstRtspServer.RTSPStreamTransport.get_rtpinfo
def GstRtspServer.RTSPStreamTransport.get_rtpinfo (self, start_time):
#python wrapper for 'gst_rtsp_stream_transport_get_rtpinfo'
Get the RTP-Info string for trans and start_time.
Parameters:
a star time
gst_rtsp_stream_transport_get_stream
GstRTSPStream * gst_rtsp_stream_transport_get_stream (GstRTSPStreamTransport * trans)
Get the GstRTSPStream used when constructing trans.
Parameters:
trans
–
the stream used when constructing trans.
GstRtspServer.RTSPStreamTransport.prototype.get_stream
function GstRtspServer.RTSPStreamTransport.prototype.get_stream(): {
// javascript wrapper for 'gst_rtsp_stream_transport_get_stream'
}
Get the GstRtspServer.RTSPStream used when constructing trans.
Parameters:
the stream used when constructing trans.
GstRtspServer.RTSPStreamTransport.get_stream
def GstRtspServer.RTSPStreamTransport.get_stream (self):
#python wrapper for 'gst_rtsp_stream_transport_get_stream'
Get the GstRtspServer.RTSPStream used when constructing trans.
Parameters:
the stream used when constructing trans.
gst_rtsp_stream_transport_get_transport
const GstRTSPTransport * gst_rtsp_stream_transport_get_transport (GstRTSPStreamTransport * trans)
Get the transport configured in trans.
Parameters:
trans
–
the transport configured in trans. It remains valid for as long as trans is valid.
GstRtspServer.RTSPStreamTransport.prototype.get_transport
function GstRtspServer.RTSPStreamTransport.prototype.get_transport(): {
// javascript wrapper for 'gst_rtsp_stream_transport_get_transport'
}
Get the transport configured in trans.
Parameters:
the transport configured in trans. It remains valid for as long as trans is valid.
GstRtspServer.RTSPStreamTransport.get_transport
def GstRtspServer.RTSPStreamTransport.get_transport (self):
#python wrapper for 'gst_rtsp_stream_transport_get_transport'
Get the transport configured in trans.
Parameters:
the transport configured in trans. It remains valid for as long as trans is valid.
gst_rtsp_stream_transport_get_url
const GstRTSPUrl * gst_rtsp_stream_transport_get_url (GstRTSPStreamTransport * trans)
Get the url configured in trans.
Parameters:
trans
–
the url configured in trans. It remains valid for as long as trans is valid.
GstRtspServer.RTSPStreamTransport.prototype.get_url
function GstRtspServer.RTSPStreamTransport.prototype.get_url(): {
// javascript wrapper for 'gst_rtsp_stream_transport_get_url'
}
Get the url configured in trans.
Parameters:
the url configured in trans. It remains valid for as long as trans is valid.
GstRtspServer.RTSPStreamTransport.get_url
def GstRtspServer.RTSPStreamTransport.get_url (self):
#python wrapper for 'gst_rtsp_stream_transport_get_url'
Get the url configured in trans.
Parameters:
the url configured in trans. It remains valid for as long as trans is valid.
gst_rtsp_stream_transport_is_timed_out
gboolean gst_rtsp_stream_transport_is_timed_out (GstRTSPStreamTransport * trans)
Check if trans is timed out.
Parameters:
trans
–
TRUE if trans timed out.
GstRtspServer.RTSPStreamTransport.prototype.is_timed_out
function GstRtspServer.RTSPStreamTransport.prototype.is_timed_out(): {
// javascript wrapper for 'gst_rtsp_stream_transport_is_timed_out'
}
Check if trans is timed out.
Parameters:
GstRtspServer.RTSPStreamTransport.is_timed_out
def GstRtspServer.RTSPStreamTransport.is_timed_out (self):
#python wrapper for 'gst_rtsp_stream_transport_is_timed_out'
Check if trans is timed out.
Parameters:
gst_rtsp_stream_transport_keep_alive
gst_rtsp_stream_transport_keep_alive (GstRTSPStreamTransport * trans)
Signal the installed keep_alive callback for trans.
Parameters:
trans
–
GstRtspServer.RTSPStreamTransport.prototype.keep_alive
function GstRtspServer.RTSPStreamTransport.prototype.keep_alive(): {
// javascript wrapper for 'gst_rtsp_stream_transport_keep_alive'
}
Signal the installed keep_alive callback for trans.
Parameters:
GstRtspServer.RTSPStreamTransport.keep_alive
def GstRtspServer.RTSPStreamTransport.keep_alive (self):
#python wrapper for 'gst_rtsp_stream_transport_keep_alive'
Signal the installed keep_alive callback for trans.
Parameters:
gst_rtsp_stream_transport_message_sent
gst_rtsp_stream_transport_message_sent (GstRTSPStreamTransport * trans)
Signal the installed message_sent / message_sent_full callback for trans.
Parameters:
trans
–
Since : 1.16
GstRtspServer.RTSPStreamTransport.prototype.message_sent
function GstRtspServer.RTSPStreamTransport.prototype.message_sent(): {
// javascript wrapper for 'gst_rtsp_stream_transport_message_sent'
}
Signal the installed message_sent / message_sent_full callback for trans.
Parameters:
Since : 1.16
GstRtspServer.RTSPStreamTransport.message_sent
def GstRtspServer.RTSPStreamTransport.message_sent (self):
#python wrapper for 'gst_rtsp_stream_transport_message_sent'
Signal the installed message_sent / message_sent_full callback for trans.
Parameters:
Since : 1.16
gst_rtsp_stream_transport_recv_data
GstFlowReturn gst_rtsp_stream_transport_recv_data (GstRTSPStreamTransport * trans, guint channel, GstBuffer * buffer)
Receive buffer on channel trans.
Parameters:
trans
–
channel
–
a channel
buffer
(
[transfer: full])
–
a GstFlowReturn. Returns GST_FLOW_NOT_LINKED when channel is not configured in the transport of trans.
GstRtspServer.RTSPStreamTransport.prototype.recv_data
function GstRtspServer.RTSPStreamTransport.prototype.recv_data(channel: Number, buffer: Gst.Buffer): {
// javascript wrapper for 'gst_rtsp_stream_transport_recv_data'
}
Receive buffer on channel trans.
Parameters:
a channel
a Gst.FlowReturn. Returns GST_FLOW_NOT_LINKED when channel is not configured in the transport of trans.
GstRtspServer.RTSPStreamTransport.recv_data
def GstRtspServer.RTSPStreamTransport.recv_data (self, channel, buffer):
#python wrapper for 'gst_rtsp_stream_transport_recv_data'
Receive buffer on channel trans.
Parameters:
a channel
a Gst.FlowReturn. Returns GST_FLOW_NOT_LINKED when channel is not configured in the transport of trans.
gst_rtsp_stream_transport_send_rtcp
gboolean gst_rtsp_stream_transport_send_rtcp (GstRTSPStreamTransport * trans, GstBuffer * buffer)
Send buffer to the installed RTCP callback for trans.
TRUE on success
GstRtspServer.RTSPStreamTransport.prototype.send_rtcp
function GstRtspServer.RTSPStreamTransport.prototype.send_rtcp(buffer: Gst.Buffer): {
// javascript wrapper for 'gst_rtsp_stream_transport_send_rtcp'
}
Send buffer to the installed RTCP callback for trans.
Parameters:
GstRtspServer.RTSPStreamTransport.send_rtcp
def GstRtspServer.RTSPStreamTransport.send_rtcp (self, buffer):
#python wrapper for 'gst_rtsp_stream_transport_send_rtcp'
Send buffer to the installed RTCP callback for trans.
Parameters:
gst_rtsp_stream_transport_send_rtcp_list
gboolean gst_rtsp_stream_transport_send_rtcp_list (GstRTSPStreamTransport * trans, GstBufferList * buffer_list)
Send buffer_list to the installed RTCP callback for trans.
TRUE on success
Since : 1.16
GstRtspServer.RTSPStreamTransport.prototype.send_rtcp_list
function GstRtspServer.RTSPStreamTransport.prototype.send_rtcp_list(buffer_list: Gst.BufferList): {
// javascript wrapper for 'gst_rtsp_stream_transport_send_rtcp_list'
}
Send buffer_list to the installed RTCP callback for trans.
Parameters:
Since : 1.16
GstRtspServer.RTSPStreamTransport.send_rtcp_list
def GstRtspServer.RTSPStreamTransport.send_rtcp_list (self, buffer_list):
#python wrapper for 'gst_rtsp_stream_transport_send_rtcp_list'
Send buffer_list to the installed RTCP callback for trans.
Parameters:
Since : 1.16
gst_rtsp_stream_transport_send_rtp
gboolean gst_rtsp_stream_transport_send_rtp (GstRTSPStreamTransport * trans, GstBuffer * buffer)
Send buffer to the installed RTP callback for trans.
TRUE on success
GstRtspServer.RTSPStreamTransport.prototype.send_rtp
function GstRtspServer.RTSPStreamTransport.prototype.send_rtp(buffer: Gst.Buffer): {
// javascript wrapper for 'gst_rtsp_stream_transport_send_rtp'
}
Send buffer to the installed RTP callback for trans.
Parameters:
GstRtspServer.RTSPStreamTransport.send_rtp
def GstRtspServer.RTSPStreamTransport.send_rtp (self, buffer):
#python wrapper for 'gst_rtsp_stream_transport_send_rtp'
Send buffer to the installed RTP callback for trans.
Parameters:
gst_rtsp_stream_transport_send_rtp_list
gboolean gst_rtsp_stream_transport_send_rtp_list (GstRTSPStreamTransport * trans, GstBufferList * buffer_list)
Send buffer_list to the installed RTP callback for trans.
TRUE on success
Since : 1.16
GstRtspServer.RTSPStreamTransport.prototype.send_rtp_list
function GstRtspServer.RTSPStreamTransport.prototype.send_rtp_list(buffer_list: Gst.BufferList): {
// javascript wrapper for 'gst_rtsp_stream_transport_send_rtp_list'
}
Send buffer_list to the installed RTP callback for trans.
Parameters:
Since : 1.16
GstRtspServer.RTSPStreamTransport.send_rtp_list
def GstRtspServer.RTSPStreamTransport.send_rtp_list (self, buffer_list):
#python wrapper for 'gst_rtsp_stream_transport_send_rtp_list'
Send buffer_list to the installed RTP callback for trans.
Parameters:
Since : 1.16
gst_rtsp_stream_transport_set_active
gboolean gst_rtsp_stream_transport_set_active (GstRTSPStreamTransport * trans, gboolean active)
Activate or deactivate datatransfer configured in trans.
TRUE when the state was changed.
GstRtspServer.RTSPStreamTransport.prototype.set_active
function GstRtspServer.RTSPStreamTransport.prototype.set_active(active: Number): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_active'
}
Activate or deactivate datatransfer configured in trans.
Parameters:
new state of trans
GstRtspServer.RTSPStreamTransport.set_active
def GstRtspServer.RTSPStreamTransport.set_active (self, active):
#python wrapper for 'gst_rtsp_stream_transport_set_active'
Activate or deactivate datatransfer configured in trans.
Parameters:
new state of trans
gst_rtsp_stream_transport_set_callbacks
gst_rtsp_stream_transport_set_callbacks (GstRTSPStreamTransport * trans, GstRTSPSendFunc send_rtp, GstRTSPSendFunc send_rtcp, gpointer user_data, GDestroyNotify notify)
Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.
Parameters:
trans
–
send_rtp
(
[scope notified][closure])
–
a callback called when RTP should be sent
send_rtcp
(
[scope notified][closure])
–
a callback called when RTCP should be sent
user_data
–
user data passed to callbacks
notify
(
[allow-none])
–
called with the user_data when no longer needed.
GstRtspServer.RTSPStreamTransport.prototype.set_callbacks
function GstRtspServer.RTSPStreamTransport.prototype.set_callbacks(send_rtp: GstRtspServer.RTSPSendFunc, send_rtcp: GstRtspServer.RTSPSendFunc, user_data: Object): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_callbacks'
}
Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.
Parameters:
a callback called when RTP should be sent
a callback called when RTCP should be sent
user data passed to callbacks
GstRtspServer.RTSPStreamTransport.set_callbacks
def GstRtspServer.RTSPStreamTransport.set_callbacks (self, send_rtp, send_rtcp, *user_data):
#python wrapper for 'gst_rtsp_stream_transport_set_callbacks'
Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.
Parameters:
a callback called when RTP should be sent
a callback called when RTCP should be sent
user data passed to callbacks
gst_rtsp_stream_transport_set_keepalive
gst_rtsp_stream_transport_set_keepalive (GstRTSPStreamTransport * trans, GstRTSPKeepAliveFunc keep_alive, gpointer user_data, GDestroyNotify notify)
Install callbacks that will be called when RTCP packets are received from the receiver of trans.
Parameters:
trans
–
keep_alive
(
[scope notified][closure])
–
a callback called when the receiver is active
user_data
–
user data passed to callback
notify
(
[allow-none])
–
called with the user_data when no longer needed.
GstRtspServer.RTSPStreamTransport.prototype.set_keepalive
function GstRtspServer.RTSPStreamTransport.prototype.set_keepalive(keep_alive: GstRtspServer.RTSPKeepAliveFunc, user_data: Object): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_keepalive'
}
Install callbacks that will be called when RTCP packets are received from the receiver of trans.
Parameters:
a callback called when the receiver is active
user data passed to callback
GstRtspServer.RTSPStreamTransport.set_keepalive
def GstRtspServer.RTSPStreamTransport.set_keepalive (self, keep_alive, *user_data):
#python wrapper for 'gst_rtsp_stream_transport_set_keepalive'
Install callbacks that will be called when RTCP packets are received from the receiver of trans.
Parameters:
a callback called when the receiver is active
user data passed to callback
gst_rtsp_stream_transport_set_list_callbacks
gst_rtsp_stream_transport_set_list_callbacks (GstRTSPStreamTransport * trans, GstRTSPSendListFunc send_rtp_list, GstRTSPSendListFunc send_rtcp_list, gpointer user_data, GDestroyNotify notify)
Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.
Parameters:
trans
–
send_rtp_list
(
[scope notified][closure])
–
a callback called when RTP should be sent
send_rtcp_list
(
[scope notified][closure])
–
a callback called when RTCP should be sent
user_data
–
user data passed to callbacks
notify
(
[allow-none])
–
called with the user_data when no longer needed.
Since : 1.16
GstRtspServer.RTSPStreamTransport.prototype.set_list_callbacks
function GstRtspServer.RTSPStreamTransport.prototype.set_list_callbacks(send_rtp_list: GstRtspServer.RTSPSendListFunc, send_rtcp_list: GstRtspServer.RTSPSendListFunc, user_data: Object): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_list_callbacks'
}
Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.
Parameters:
a callback called when RTP should be sent
a callback called when RTCP should be sent
user data passed to callbacks
Since : 1.16
GstRtspServer.RTSPStreamTransport.set_list_callbacks
def GstRtspServer.RTSPStreamTransport.set_list_callbacks (self, send_rtp_list, send_rtcp_list, *user_data):
#python wrapper for 'gst_rtsp_stream_transport_set_list_callbacks'
Install callbacks that will be called when data for a stream should be sent to a client. This is usually used when sending RTP/RTCP over TCP.
Parameters:
a callback called when RTP should be sent
a callback called when RTCP should be sent
user data passed to callbacks
Since : 1.16
gst_rtsp_stream_transport_set_message_sent
gst_rtsp_stream_transport_set_message_sent (GstRTSPStreamTransport * trans, GstRTSPMessageSentFunc message_sent, gpointer user_data, GDestroyNotify notify)
Install a callback that will be called when a message has been sent on trans.
Parameters:
trans
–
message_sent
(
[scope notified][closure])
–
a callback called when a message has been sent
user_data
–
user data passed to callback
notify
(
[allow-none])
–
called with the user_data when no longer needed
GstRtspServer.RTSPStreamTransport.prototype.set_message_sent
function GstRtspServer.RTSPStreamTransport.prototype.set_message_sent(message_sent: GstRtspServer.RTSPMessageSentFunc, user_data: Object): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_message_sent'
}
Install a callback that will be called when a message has been sent on trans.
Parameters:
a callback called when a message has been sent
user data passed to callback
GstRtspServer.RTSPStreamTransport.set_message_sent
def GstRtspServer.RTSPStreamTransport.set_message_sent (self, message_sent, *user_data):
#python wrapper for 'gst_rtsp_stream_transport_set_message_sent'
Install a callback that will be called when a message has been sent on trans.
Parameters:
a callback called when a message has been sent
user data passed to callback
gst_rtsp_stream_transport_set_message_sent_full
gst_rtsp_stream_transport_set_message_sent_full (GstRTSPStreamTransport * trans, GstRTSPMessageSentFuncFull message_sent, gpointer user_data, GDestroyNotify notify)
Install a callback that will be called when a message has been sent on trans.
Parameters:
trans
–
message_sent
(
[scope notified][closure])
–
a callback called when a message has been sent
user_data
–
user data passed to callback
notify
(
[allow-none])
–
called with the user_data when no longer needed
Since : 1.18
GstRtspServer.RTSPStreamTransport.prototype.set_message_sent_full
function GstRtspServer.RTSPStreamTransport.prototype.set_message_sent_full(message_sent: GstRtspServer.RTSPMessageSentFuncFull, user_data: Object): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_message_sent_full'
}
Install a callback that will be called when a message has been sent on trans.
Parameters:
a callback called when a message has been sent
user data passed to callback
Since : 1.18
GstRtspServer.RTSPStreamTransport.set_message_sent_full
def GstRtspServer.RTSPStreamTransport.set_message_sent_full (self, message_sent, *user_data):
#python wrapper for 'gst_rtsp_stream_transport_set_message_sent_full'
Install a callback that will be called when a message has been sent on trans.
Parameters:
a callback called when a message has been sent
user data passed to callback
Since : 1.18
gst_rtsp_stream_transport_set_timed_out
gst_rtsp_stream_transport_set_timed_out (GstRTSPStreamTransport * trans, gboolean timedout)
Set the timed out state of trans to timedout
GstRtspServer.RTSPStreamTransport.prototype.set_timed_out
function GstRtspServer.RTSPStreamTransport.prototype.set_timed_out(timedout: Number): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_timed_out'
}
Set the timed out state of trans to timedout
Parameters:
timed out value
GstRtspServer.RTSPStreamTransport.set_timed_out
def GstRtspServer.RTSPStreamTransport.set_timed_out (self, timedout):
#python wrapper for 'gst_rtsp_stream_transport_set_timed_out'
Set the timed out state of trans to timedout
Parameters:
timed out value
gst_rtsp_stream_transport_set_transport
gst_rtsp_stream_transport_set_transport (GstRTSPStreamTransport * trans, GstRTSPTransport * tr)
Set tr as the client transport. This function takes ownership of the passed tr.
GstRtspServer.RTSPStreamTransport.prototype.set_transport
function GstRtspServer.RTSPStreamTransport.prototype.set_transport(tr: GstRtsp.RTSPTransport): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_transport'
}
Set tr as the client transport. This function takes ownership of the passed tr.
Parameters:
a client GstRtsp.RTSPTransport
GstRtspServer.RTSPStreamTransport.set_transport
def GstRtspServer.RTSPStreamTransport.set_transport (self, tr):
#python wrapper for 'gst_rtsp_stream_transport_set_transport'
Set tr as the client transport. This function takes ownership of the passed tr.
Parameters:
a client GstRtsp.RTSPTransport
gst_rtsp_stream_transport_set_url
gst_rtsp_stream_transport_set_url (GstRTSPStreamTransport * trans, const GstRTSPUrl * url)
Set url as the client url.
GstRtspServer.RTSPStreamTransport.prototype.set_url
function GstRtspServer.RTSPStreamTransport.prototype.set_url(url: GstRtsp.RTSPUrl): {
// javascript wrapper for 'gst_rtsp_stream_transport_set_url'
}
Set url as the client url.
Parameters:
a client GstRtsp.RTSPUrl
GstRtspServer.RTSPStreamTransport.set_url
def GstRtspServer.RTSPStreamTransport.set_url (self, url):
#python wrapper for 'gst_rtsp_stream_transport_set_url'
Set url as the client url.
Parameters:
a client GstRtsp.RTSPUrl
Function Macros
GST_RTSP_STREAM_TRANSPORT_CAST
#define GST_RTSP_STREAM_TRANSPORT_CAST(obj) ((GstRTSPStreamTransport*)(obj))
GST_RTSP_STREAM_TRANSPORT_CLASS_CAST
#define GST_RTSP_STREAM_TRANSPORT_CLASS_CAST(klass) ((GstRTSPStreamTransportClass*)(klass))
Callbacks
GstRTSPKeepAliveFunc
(*GstRTSPKeepAliveFunc) (gpointer user_data)
Function registered with gst_rtsp_stream_transport_set_keepalive and called when the stream is active.
Parameters:
user_data
–
user data
GstRtspServer.RTSPKeepAliveFunc
function GstRtspServer.RTSPKeepAliveFunc(user_data: Object): {
// javascript wrapper for 'GstRTSPKeepAliveFunc'
}
Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_keepalive and called when the stream is active.
Parameters:
user data
GstRtspServer.RTSPKeepAliveFunc
def GstRtspServer.RTSPKeepAliveFunc (*user_data):
#python wrapper for 'GstRTSPKeepAliveFunc'
Function registered with GstRtspServer.RTSPStreamTransport.set_keepalive and called when the stream is active.
Parameters:
user data
GstRTSPMessageSentFunc
(*GstRTSPMessageSentFunc) (gpointer user_data)
Function registered with gst_rtsp_stream_transport_set_message_sent and called when a message has been sent on the transport.
Parameters:
user_data
–
user data
GstRtspServer.RTSPMessageSentFunc
function GstRtspServer.RTSPMessageSentFunc(user_data: Object): {
// javascript wrapper for 'GstRTSPMessageSentFunc'
}
Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_message_sent and called when a message has been sent on the transport.
Parameters:
user data
GstRtspServer.RTSPMessageSentFunc
def GstRtspServer.RTSPMessageSentFunc (*user_data):
#python wrapper for 'GstRTSPMessageSentFunc'
Function registered with GstRtspServer.RTSPStreamTransport.set_message_sent and called when a message has been sent on the transport.
Parameters:
user data
GstRTSPMessageSentFuncFull
(*GstRTSPMessageSentFuncFull) (GstRTSPStreamTransport * trans, gpointer user_data)
Function registered with gst_rtsp_stream_transport_set_message_sent_full and called when a message has been sent on the transport.
Parameters:
trans
–
user_data
–
user data
Since : 1.18
GstRtspServer.RTSPMessageSentFuncFull
function GstRtspServer.RTSPMessageSentFuncFull(trans: GstRtspServer.RTSPStreamTransport, user_data: Object): {
// javascript wrapper for 'GstRTSPMessageSentFuncFull'
}
Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_message_sent_full and called when a message has been sent on the transport.
Parameters:
user data
Since : 1.18
GstRtspServer.RTSPMessageSentFuncFull
def GstRtspServer.RTSPMessageSentFuncFull (trans, *user_data):
#python wrapper for 'GstRTSPMessageSentFuncFull'
Function registered with GstRtspServer.RTSPStreamTransport.set_message_sent_full and called when a message has been sent on the transport.
Parameters:
user data
Since : 1.18
GstRTSPSendFunc
gboolean (*GstRTSPSendFunc) (GstBuffer * buffer, guint8 channel, gpointer user_data)
Function registered with gst_rtsp_stream_transport_set_callbacks and called when buffer must be sent on channel.
TRUE on success
GstRtspServer.RTSPSendFunc
function GstRtspServer.RTSPSendFunc(buffer: Gst.Buffer, channel: Number, user_data: Object): {
// javascript wrapper for 'GstRTSPSendFunc'
}
Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_callbacks and called when buffer must be sent on channel.
Parameters:
a channel
user data
GstRtspServer.RTSPSendFunc
def GstRtspServer.RTSPSendFunc (buffer, channel, *user_data):
#python wrapper for 'GstRTSPSendFunc'
Function registered with GstRtspServer.RTSPStreamTransport.set_callbacks and called when buffer must be sent on channel.
Parameters:
a channel
user data
GstRTSPSendListFunc
gboolean (*GstRTSPSendListFunc) (GstBufferList * buffer_list, guint8 channel, gpointer user_data)
Function registered with gst_rtsp_stream_transport_set_callbacks and called when buffer_list must be sent on channel.
TRUE on success
Since : 1.16
GstRtspServer.RTSPSendListFunc
function GstRtspServer.RTSPSendListFunc(buffer_list: Gst.BufferList, channel: Number, user_data: Object): {
// javascript wrapper for 'GstRTSPSendListFunc'
}
Function registered with GstRtspServer.RTSPStreamTransport.prototype.set_callbacks and called when buffer_list must be sent on channel.
Parameters:
a channel
user data
Since : 1.16
GstRtspServer.RTSPSendListFunc
def GstRtspServer.RTSPSendListFunc (buffer_list, channel, *user_data):
#python wrapper for 'GstRTSPSendListFunc'
Function registered with GstRtspServer.RTSPStreamTransport.set_callbacks and called when buffer_list must be sent on channel.
Parameters:
a channel
user data
Since : 1.16
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