gstreamer/doc/rtsp/rtsp-server2.md
# rtsp-server 源码分析
```c
[rtsp-server]
gst_rtsp_server_attach (GstRTSPServer * server, GMainContext * context)
//create a source from server and attach the source to the context
|
|
↓
gst_rtsp_server_create_source(GstRTSPServer * server, GCancellable * cancellable, GError ** error)
// listening for connections on a port and creating #GstRTSPClient objects to handle those connections.
// When a new connection is received, a new #GstRTSPClient object is created to handle the connection.
* gst_rtsp_server_create_socket()
* g_socket_create_source() /* create a watch for reads (new connections) and possible errors */
* g_source_set_callback()--->gst_rtsp_server_io_func()
||
||
↓↓
gst_rtsp_server_io_func(GSocket * socket, GIOCondition condition, GstRTSPServer * server)
// creates a new #GstRTSPClient to accept and handle a new connection on @socket or @server.
* create_client (server);--->default_create_client()--->[rtsp-client]gst_rtsp_client_new()
* [rtsp-client] gst_rtsp_client_set_connection (client, conn);
//log
** Message: [rtsp-client] client 000001C62478F110 connected to server ip 127.0.0.1, ipv6 = 0
** Message: [rtsp-client] added new client 000001C62478F110 ip 127.0.0.1:57323
* manage_client (server, client);/* add the client context to the active list of clients, takes ownership of client */
| //signal
| SIGNAL_CLIENT_CONNECTED
| g_signal_connect (client, "closed", (GCallback) unmanage_client, cctx);
╰──╮
↓
gst_rtsp_client_attach()--->gst_rtsp_watch_new()--->watch_funcs()--->message_received()
--->message_sent()
...
[rtsp-client]
message_received()--->gst_rtsp_client_handle_message()--->handle_request()
//DESCRIBE
handle_describe_request()
| //signal
| SIGNAL_PRE_DESCRIBE_REQUEST
| SIGNAL_DESCRIBE_REQUEST
╰──╮
↓
find_media()
--->[rtsp-media-factory]gst_rtsp_media_factory_construct()/*create #rtspmedia*/
| //signal
| SIGNAL_MEDIA_CONSTRUCTED
| SIGNAL_MEDIA_CONFIGURE
╰──>[rtsp-media-factory]default_construct()
╰──>[rtsp-media-factory]gst_rtsp_media_factory_create_element()/*create #element according launch*/★
╰──>[rtsp-media]gst_rtsp_media_collect_streams()
╰──>[rtsp-media]gst_rtsp_media_create_stream(media, elem, pad)/*create #rtspstream*/★
╰──>[rtsp-media-factory]default_create_pipeline()
pipeline = gst_pipeline_new ("media-pipeline");
╰──>[rtsp-media]gst_rtsp_media_take_pipeline(media,pipeline);/*create #rtspstream #pipeline and add #element into #pipeline*/★
--->[rtsp-media]gst_rtsp_media_prepare()/*The pipeline of the media is set to PAUSED*/
| //status
| GST_RTSP_MEDIA_STATUS_PREPARING
╰──>[rtsp-media]default_prepare()
╰──>[rtsp-media]create_rtpbin()/*create #rtpbin*/
╰──>[rtsp-media]gst_bin_add (GST_BIN (priv->pipeline), priv->rtpbin);/*add rtpbin to rtpsmedia pipeline*/★
╰──>[rtsp-media]start_prepare()
╰──>[rtsp-stream]gst_rtsp_stream_join_bin()/*join stream to #rtpbin*/
╰──>[rtsp-stream]create_sender_part(stream, bin, state);
╰──>[rtsp-stream]create_receiver_part(stream, bin, state);
╰──>[rtsp-media]start_preroll()/*set pipeline GST_STATE_PAUSED*/
╰──>[rtsp-media]media_streams_set_blocked()/* start blocked to make sure nothing goes to the sink */
╰──>[rtsp-media]set_state (media, GST_STATE_PLAYING);
//signal
SIGNAL_PREPARED
### media->priv->pipeline
包含整条流的pipeline ，可以通过gst_rtsp_media_take_pipeline()设置
### media->priv->element
// create by gst_rtsp_media_new(), it will provide the streaming elements.
gst_rtsp_media_factory_set_launch()里面的根据语法解析出来的element会添加到[media->priv->element]里面
media->priv->element可以看作是一个bin
// For each of the streams, a new GstRTSPStream object needs to be made with the gst_rtsp_media_create_stream()
gst_rtsp_media_create_stream(media, pay, pad)
* pay为media->priv->element中提取出的payload element，也就是"xxxpay"这样的element。
* pad为elem的src pad
1. 根据pad创建一个ghost_pad并添加到media->priv->element中去。
2. 创建stream（GstElement），`stream会和ghost_pad连接`。
stream = gst_rtsp_stream_new (idx, payloader, ghostpad);
`stream->priv->srcpad = gst_object_ref (ghostpad);`
3. 创建的stream添加到priv->streams这个数组中去。
gst_rtsp_media_take_pipeline()--->gst_bin_add (GST_BIN_CAST (pipeline), priv->element);
gst_rtsp_media_set_property
[rtsp-client]handle_play_request-->[rtsp-media]gst_rtsp_media_complete_pipeline-->[rtsp-stream]gst_rtsp_stream_complete_stream-->[rtsp-stream]create_sender_part
## rtsp-stream
gst_rtsp_stream_join_bin()旧版
1. stream的`srcpad`和rtpbin的send_rtp_sink pad连接起来，**也即将element与rtpbin连起来**
2. create_sender_part (stream, bin, state);
这个bin就是priv->pipeline，这里会创建tee和用于发rtp/rtcp的udpsink或者appsink，然后把tee和rtpbin连接起来
3. create_receiver_part (stream, bin, state);
create_sender_part
/* For the sender we create this bit of pipeline for both
* RTP and RTCP. Sync and preroll are enabled on udpsink so
* we need to add a queue before appsink and udpsink to make
* the pipeline not block. For the TCP case, we want to pump
* client as fast as possible anyway. This pipeline is used
* when both TCP and UDP are present.
*
* .--------. .-----. .---------. .---------.
* | rtpbin | | tee | | queue | | udpsink |
* | send->sink src->sink src->sink |
* '--------' | | '---------' '---------'
* | | .---------. .---------.
* | | | queue | | appsink |
* | src->sink src->sink |
* '-----' '---------' '---------'
*
* When only UDP or only TCP is allowed, we skip the tee and queue
* and link the udpsink (for UDP) or appsink (for TCP) directly to
* the session.
*/
create_receiver_part
/* For the receiver we create this bit of pipeline for both
* RTP and RTCP. We receive RTP/RTCP on appsrc and udpsrc
* and it is all funneled into the rtpbin receive pad.
*
* .--------. .--------. .--------.
* | udpsrc | | funnel | | rtpbin |
* | src->sink src->sink |
* '--------' | | '--------'
* .--------. | |
* | appsrc | | |
* | src->sink |
* '--------' '--------'
*/
typedef enum {
GST_RTSP_MEDIA_STATUS_UNPREPARED = 0,
GST_RTSP_MEDIA_STATUS_UNPREPARING = 1,
GST_RTSP_MEDIA_STATUS_PREPARING = 2,
GST_RTSP_MEDIA_STATUS_PREPARED = 3,
GST_RTSP_MEDIA_STATUS_SUSPENDED = 4,
GST_RTSP_MEDIA_STATUS_ERROR = 5
} GstRTSPMediaStatus;
```