mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-28 11:10:37 +00:00
docs: update docs and comments
This commit is contained in:
parent
92eb244215
commit
996112db95
3 changed files with 130 additions and 49 deletions
174
docs/README
174
docs/README
|
@ -103,7 +103,7 @@ can build simple server applications with it.
|
|||
alternative implementation can be used by the server.
|
||||
|
||||
The GstRTSPMediaMapping object is more interesting and needs more configuration
|
||||
before the server object is useful. This object manages to mapping from a
|
||||
before the server object is useful. This object manages the mapping from a
|
||||
request URL to a specific stream and its configuration. We explain in the next
|
||||
topic how to configure this object.
|
||||
|
||||
|
@ -202,7 +202,7 @@ can build simple server applications with it.
|
|||
have to be negotiated with the client in the SETUP requests.
|
||||
|
||||
When preparing a GstRTSPMedia, a multifdsink is also constructed for streaming
|
||||
the stream over TCP^when requested.
|
||||
the stream over TCP when requested.
|
||||
|
||||
|
||||
* the GstRTSPClient object
|
||||
|
@ -212,61 +212,143 @@ can build simple server applications with it.
|
|||
a new GstRTCPClient object, will configure the session pool and media mapper
|
||||
objects in it and will then call the accept function of the client.
|
||||
|
||||
The default GstRTSPClient will accept the connection and will start a new
|
||||
GThread to handle the connection. In RTSP it is usual to keep the connection
|
||||
open between multiple RTSP requests. The client thread will simply block for a
|
||||
new GstRTSPMessage, will dispatch it and will send a response.
|
||||
The default GstRTSPClient will accept the connection and will attach a watch to
|
||||
the server mainloop. In RTSP it is usual to keep the connection
|
||||
open between multiple RTSP requests. The client watch will be dispatched by the
|
||||
server mainloop when a new GstRTSPMessage is received, which will then be
|
||||
handled and a response will be sent.
|
||||
|
||||
We will briefly describe how it deals with some common requests.
|
||||
|
||||
- DESCRIBE:
|
||||
|
||||
locates the GstRTSPMedia for the url, prepares it and asks the sdp helper
|
||||
function to construct an SDP from the caps of the prepared media pipeline.
|
||||
It will also cache the url+media object so that it can be reused later.
|
||||
|
||||
- SETUP
|
||||
|
||||
A new GstRTSPSession object will be created from the GstRTSPSessionPool
|
||||
object configured in the GstRTSPClient. This session will contain the
|
||||
configuration of the client regarding the media it is streaming and the
|
||||
ports/transport it negotiated with the server.
|
||||
|
||||
The sessionid is set in the response header. The client will add the
|
||||
sessionid to any further SETUP/PLAY/PAUSE/TEARDOWN request so that we can
|
||||
always find the session again.
|
||||
|
||||
The session configuration for a sessionid will have a link to the prepared
|
||||
GstRTSPMedia object of the stream. The port and transport of the client is
|
||||
stored in the session configuration.
|
||||
|
||||
- PLAY
|
||||
|
||||
The session configuration is retrieved with the sessionid and the client
|
||||
ports are configured in the UDP sinks, then the streaming to the client
|
||||
is started.
|
||||
|
||||
- PAUSE
|
||||
|
||||
The session configuration is retrieved with the sessionid and the client
|
||||
ports are removed from the UDP sinks, the streaming to the client
|
||||
pauses.
|
||||
|
||||
- TEARDOWN
|
||||
|
||||
The session configuration is released along with its link to the
|
||||
GstRTSPMedia object. When no more clients are refering to the GstRTSPMedia
|
||||
object, it can be released as well.
|
||||
The GstRTSPClient object remains alive for as long as a client has a TCP
|
||||
connection open with the server. Since is possible for a client to open and close
|
||||
the TCP connection between requests, we cannot store the state related
|
||||
to the configured RTSP session in the GstRTSPClient object. This server state
|
||||
is instead stored in the GstRTSPSession object.
|
||||
|
||||
|
||||
* GstRTSPSession
|
||||
|
||||
This object contains state about a specific RTSP session identified with a
|
||||
session id. This state contains the configured streams and their associated
|
||||
transports.
|
||||
|
||||
When a GstRTSPClient performs a SETUP request, the server will allocate a new
|
||||
GstRTSPSession with a unique session id from the GstRTSPSessionPool. The pool
|
||||
maintains a list of all existing sessions and makes sure that no session id is
|
||||
used multiple times. The session id is sent to the client so that the client
|
||||
can refer to its previously configured state by sending the session id in
|
||||
further requests.
|
||||
|
||||
A client will then use the session id to configure one or more streams,
|
||||
identified by their url. This information is kept in a GstRTSPSessionMedia
|
||||
structure that is refered to from the GstRTSPSession.
|
||||
|
||||
|
||||
* GstRTSPSessionMedia and GstRTSPSessionStream
|
||||
|
||||
A GstRTSPSessionMedia is identified by a URL and is referenced by a
|
||||
GstRTSPSession. It is created as soon as a client performs a SETUP operation on
|
||||
a particular URL. It will contain a link to the GstRTSPMedia object associated
|
||||
with the URL along with the state of the media and the configured transports
|
||||
for each of the streams in the media.
|
||||
|
||||
Each SETUP request performed by the client will configure a
|
||||
GstRTSPSessionStream object linked to by the GstRTSPSessionMedia structure.
|
||||
It will contain the transport information needed to send this stream to the
|
||||
client. The GstRTSPSessionStream also contains a link to the GstRTSPMediaStream
|
||||
object that generates the actual data to be streamed to the client.
|
||||
|
||||
Note how GstRTSPMedia and GstRTSPMediaStream (the providers of the data to
|
||||
stream) are decoupled from GstRTSPSessionMedia and GstRTSPSessionStream (the
|
||||
configuration of how to send this stream to a client) in order to be able to
|
||||
send the data of one GstRTSPMedia to multiple clients.
|
||||
|
||||
|
||||
* media control
|
||||
|
||||
After a client has configured the transports for a GstRTSPMedia and its
|
||||
GstRTSPMediaStreams, the client can play/pause/stop the stream.
|
||||
|
||||
The GstRTSPMedia object was prepared in the DESCRIBE call (or during SETUP when
|
||||
the client skipped the DESCRIBE request). As seen earlier, this configures a
|
||||
couple of multiudpsink and udpsrc elements to respectively send and receive the
|
||||
media to clients.
|
||||
|
||||
When a client performs a PLAY request, its configured destination UDP ports are
|
||||
added to the GstRTSPMediaStream target destinations, at which point data will
|
||||
be sent to the client. The corresponding GstRTSPMedia object will be set to the
|
||||
PLAYING state if it was not allready in order to send the data to the
|
||||
destination.
|
||||
|
||||
The server needs to prepare an RTP-Info header field in the PLAY response,
|
||||
which consists of the sequence number and the RTP timestamp of the next RTP
|
||||
packet. In order to achive this, the server queries the payloaders for this
|
||||
information when it prerolled the pipeline.
|
||||
|
||||
When a client performs a PAUSE request, the destination UDP ports are removed
|
||||
from the GstRTSPMediaStream object and the GstRTSPMedia object is set to PAUSED
|
||||
if no other destinations are configured anymore.
|
||||
|
||||
|
||||
* seeking
|
||||
|
||||
A seek is performed when a client sends a Range header in the PLAY request.
|
||||
This only works when not dealing with shared (live) streams.
|
||||
|
||||
The server performs a GStreamer flushing seek on the media, waits for the
|
||||
pipeline to preroll again and then responds to the client after collecting the
|
||||
new RTP sequence number and timestamp from the payloaders.
|
||||
|
||||
|
||||
* session management
|
||||
|
||||
The server has to react to clients that suddenly disappear because of network
|
||||
problems or otherwise. It needs to make sure that it can reasonable free the
|
||||
resources that are used by the various objects in use for streaming when the
|
||||
client appears to be gone.
|
||||
|
||||
Each of the GstRTSPSession objects managed by a GstRTSPSessionPool has
|
||||
therefore a last_access field that contains the timestamp of when activity from
|
||||
a client was last recorded.
|
||||
|
||||
Various ways exist to detect activity from a client:
|
||||
|
||||
- RTSP keepalive requests. When a client is receiving RTP data, the RTSP TCP
|
||||
connection is largely unused. It is the client responsability to
|
||||
periodically send keep-alive requests over the TCP channel.
|
||||
|
||||
Whenever a keep-alive request is received by the server (any request that
|
||||
contains a session id, usually an OPTION or GET_PARAMETER request) the
|
||||
last_access of the session is updated.
|
||||
|
||||
- Since it is not required for a client to keep the RTSP TCP connection open
|
||||
while streaming, gst-rtsp-server also detects activity from clients by
|
||||
looking at the RTCP messages it receives.
|
||||
|
||||
When an RTCP message is received from a client, the server looks in its list
|
||||
of active ports if this message originates from a known host/port pair that
|
||||
is currently active in a GstRTSPSession. If this is the case, the session is
|
||||
kept alive.
|
||||
|
||||
Since the server does not know anything about the port number that will be
|
||||
used by the client to send RTCP, this method does not always work. Later
|
||||
RTSP RFCs will include support for negotiating this port number with the
|
||||
server. Most clients however use the same port number for sending and
|
||||
receiving RTCP exactly for this reason.
|
||||
|
||||
If there was no activity in a particular session for a long time (by default 60
|
||||
seconds), the sessionpool will destroy the session along with all related
|
||||
objects and media as if a TEARDOWN happened from the client.
|
||||
|
||||
|
||||
* TEARDOWN
|
||||
|
||||
A TEARDOWN request will first location the GstRTSPSessionMedia of the URL. It
|
||||
will then remove all transports from the streams, making sure that streaming
|
||||
stops to the client. It will then remove the GstRTSPSessionMedia and
|
||||
GstRTSPSessionStream structures. Finally the GstRTSPSession is released back
|
||||
into the pool.
|
||||
|
||||
When there are no more references to the GstRTSPMedia, the media pipeline is
|
||||
shut down and destroyed.
|
||||
|
||||
|
||||
|
|
|
@ -1475,7 +1475,7 @@ static GstRTSPWatchFuncs watch_funcs = {
|
|||
* @client: a #GstRTSPClient
|
||||
* @channel: a #GIOChannel
|
||||
*
|
||||
* Accept a new connection for @client on the socket in @source.
|
||||
* Accept a new connection for @client on the socket in @channel.
|
||||
*
|
||||
* This function should be called when the client properties and urls are fully
|
||||
* configured and the client is ready to start.
|
||||
|
|
|
@ -464,8 +464,7 @@ default_accept_client (GstRTSPServer *server, GIOChannel *channel)
|
|||
|
||||
/* set the session pool that this client should use */
|
||||
gst_rtsp_client_set_session_pool (client, server->session_pool);
|
||||
|
||||
/* set the session pool that this client should use */
|
||||
/* set the media mapping that this client should use */
|
||||
gst_rtsp_client_set_media_mapping (client, server->media_mapping);
|
||||
|
||||
/* accept connections for that client, this function returns after accepting
|
||||
|
|
Loading…
Reference in a new issue