Free a %NULL-terminated array of credentials returned from gst_rtsp_message_parse_auth_credentials().
a %NULL-terminated array of #GstRTSPAuthCredential
Accept a new connection on socket and create a new #GstRTSPConnection for
handling communication on new socket.
a socket
a #GCancellable to cancel the operation
Create a newly allocated #GstRTSPConnection from url and store it in conn.
The connection will not yet attempt to connect to url, use
gst_rtsp_connection_connect().
A copy of url will be made.
a #GstRTSPUrl
Create a new #GstRTSPConnection for handling communication on the existing
socket socket. The initial_buffer contains zero terminated data already
read from socket which should be used before starting to read new data.
a #GSocket
the IP address of the other end
the port used by the other end
data already read from fd
Convert header to a #GstRTSPHeaderField.
a header string
Convert method to a #GstRTSPMethod.
a method
Calculates the digest auth response from the values given by the server and the username and password. See RFC2069 for details.
Currently only supported algorithm "md5".
Hash algorithm to use, or %NULL for MD5
Request method, e.g. PLAY
Realm
Username
Password
Original request URI
Nonce
Calculates the digest auth response from the values given by the server and the md5sum. See RFC2069 for details.
This function is useful when the passwords are not stored in clear text, but instead in the same format as the .htdigest file.
Currently only supported algorithm "md5".
Hash algorithm to use, or %NULL for MD5
Request method, e.g. PLAY
The md5 sum of username:realm:password
Original request URI
Nonce
Check whether field may appear multiple times in a message.
a #GstRTSPHeaderField
Convert field to a string.
a #GstRTSPHeaderField
Create a new initialized #GstRTSPMessage. Free with gst_rtsp_message_free().
Create a new data #GstRTSPMessage with channel and store the
result message in msg. Free with gst_rtsp_message_free().
the channel
Create a new #GstRTSPMessage with method and uri and store the result
request message in msg. Free with gst_rtsp_message_free().
the request method to use
the uri of the request
Create a new response #GstRTSPMessage with code and reason and store the
result message in msg. Free with gst_rtsp_message_free().
When reason is %NULL, the default reason for code will be used.
When request is not %NULL, the relevant headers will be copied to the new
response message.
the status code
the status reason or %NULL
the request that triggered the response or %NULL
Convert method to a string.
a #GstRTSPMethod
Convert options to a string.
one or more #GstRTSPMethod
Convert the comma separated list options to a #GstRTSPMethod bitwise or
of methods. This functions is the reverse of gst_rtsp_options_as_text().
a comma separated list of options
Converts the range in-place between different types of units. Ranges containing the special value #GST_RTSP_TIME_NOW can not be converted as these are only valid for #GST_RTSP_RANGE_NPT.
a #GstRTSPTimeRange
the unit to convert the range into
Free the memory allocated by range.
a #GstRTSPTimeRange
Retrieve the minimum and maximum values from range converted to
#GstClockTime in min and max.
A value of %GST_CLOCK_TIME_NONE will be used to signal #GST_RTSP_TIME_NOW
and #GST_RTSP_TIME_END for min and max respectively.
UTC times will be converted to nanoseconds since 1900.
a #GstRTSPTimeRange
Parse rangestr to a #GstRTSPTimeRange.
a range string to parse
Convert range into a string representation.
a #GstRTSPTimeRange
Convert code to a string.
a #GstRTSPStatusCode
Convert result in a human readable string.
a #GstRTSPResult
Get the #GstElement that can handle the buffers transported over trans.
It is possible that there are several managers available, use option to
selected one.
manager will contain an element name or %NULL when no manager is
needed/available for trans.
a #GstRTSPTransMode
option index.
Get the mime type of the transport mode trans. This mime type is typically
used to generate #GstCaps events.
a #GstRTSPTransMode
location to hold the result
Allocate a new initialized #GstRTSPTransport. Use gst_rtsp_transport_free() after usage.
location to hold the new #GstRTSPTransport
Parse the RTSP transport string str into transport.
a transport string
a #GstRTSPTransport
Parse the RTSP urlstr into a newly allocated #GstRTSPUrl. Free after usage
with gst_rtsp_url_free().
the url string to parse
Convert version to a string.
a #GstRTSPVersion
The default RTSP port to connect to.