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gst/rtsp/gstrtspsrc.*: Parse server address from SDP.
Original commit message from CVS: * gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize), (gst_rtspsrc_stream_free), (request_pt_map), (gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open): * gst/rtsp/gstrtspsrc.h: Parse server address from SDP. Hook up a udpsink to send RTCP back to the server. * docs/plugins/gst-plugins-good-plugins-sections.txt: * gst/rtsp/rtsptransport.h: Add some docs.
This commit is contained in:
Wim Taymans
parent
fa7454bda2
commit
1beeda3ff2
5 changed files with 110 additions and 2 deletions
13
ChangeLog
13
ChangeLog
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@ -1,3 +1,16 @@
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2007-04-25 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
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(gst_rtspsrc_stream_free), (request_pt_map),
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(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
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* gst/rtsp/gstrtspsrc.h:
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Parse server address from SDP.
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Hook up a udpsink to send RTCP back to the server.
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* gst/rtsp/rtsptransport.h:
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Add some docs.
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2007-04-25 Stefan Kost <ensonic@users.sf.net>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
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@ -489,6 +489,7 @@ gst_progress_report_get_type
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<SECTION>
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<FILE>element-rtspsrc</FILE>
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RTSPLowerTrans
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GstRTSPSrc
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<TITLE>rtspsrc</TITLE>
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<SUBSECTION Standard>
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@ -311,6 +311,7 @@ gst_rtspsrc_finalize (GObject * object)
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g_free (rtspsrc->req_location);
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g_free (rtspsrc->content_base);
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rtsp_url_free (rtspsrc->url);
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g_free (rtspsrc->addr);
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if (rtspsrc->extension) {
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#ifdef WITH_EXT_REAL
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@ -556,6 +557,12 @@ gst_rtspsrc_stream_free (GstRTSPSrc * src, GstRTSPStream * stream)
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stream->udpsrc[i] = NULL;
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}
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}
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if (stream->udpsink) {
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gst_element_set_state (stream->udpsink, GST_STATE_NULL);
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gst_bin_remove (GST_BIN_CAST (src), stream->udpsink);
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gst_object_unref (stream->udpsink);
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stream->udpsink = NULL;
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}
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if (stream->srcpad) {
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gst_pad_set_active (stream->srcpad, FALSE);
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if (stream->added) {
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@ -1086,6 +1093,8 @@ request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src)
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GstRTSPStream *stream;
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GList *lstream;
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GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
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lstream = g_list_find_custom (src->streams, GINT_TO_POINTER (session),
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(GCompareFunc) find_stream_by_id);
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if (!lstream)
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@ -1093,8 +1102,6 @@ request_pt_map (GstElement * sess, guint session, guint pt, GstRTSPSrc * src)
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stream = (GstRTSPStream *) lstream->data;
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GST_DEBUG_OBJECT (src, "getting pt map for pt %d in session %d", pt, session);
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return stream->caps;
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unknown_stream:
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@ -1312,6 +1319,38 @@ use_no_manager:
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gst_object_unref (pad);
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}
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}
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/* configure udpsink back to the server for RTCP messages. */
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{
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GstPad *pad;
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stream->udpsink = gst_element_factory_make ("udpsink", NULL);
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if (stream->udpsink == NULL)
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goto no_sink_element;
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/* we keep this playing always */
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gst_element_set_locked_state (stream->udpsink, TRUE);
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gst_element_set_state (stream->udpsink, GST_STATE_PLAYING);
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/* no sync needed */
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g_object_set (G_OBJECT (stream->udpsink), "sync", FALSE, NULL);
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/* configure host and port */
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g_object_set (G_OBJECT (stream->udpsink), "host", src->addr, "port",
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transport->server_port.max, NULL);
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gst_object_ref (stream->udpsink);
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gst_bin_add (GST_BIN_CAST (src), stream->udpsink);
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stream->rtcppad = gst_element_get_pad (stream->udpsink, "sink");
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/* get session RTCP pad */
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name = g_strdup_printf ("rtcp_src_%d", stream->id);
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pad = gst_element_get_request_pad (src->session, name);
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g_free (name);
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/* and link */
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gst_pad_link (pad, stream->rtcppad);
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}
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}
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if (src->session && !src->session_sig_id) {
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@ -1365,6 +1404,11 @@ no_element:
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GST_DEBUG_OBJECT (src, "no UDP source element found");
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return FALSE;
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}
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no_sink_element:
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{
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GST_DEBUG_OBJECT (src, "no UDP sink element found");
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return FALSE;
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}
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start_session_failure:
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{
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GST_DEBUG_OBJECT (src, "could not start session");
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@ -2570,6 +2614,16 @@ gst_rtspsrc_open (GstRTSPSrc * src)
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if (src->extension && src->extension->parse_sdp)
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src->extension->parse_sdp (src->extension, &sdp);
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/* parse address */
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{
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SDPOrigin *origin;
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origin = sdp_message_get_origin (&sdp);
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g_free (src->addr);
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src->addr = g_strdup (origin->addr);
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}
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/* create streams */
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n_streams = sdp_message_medias_len (&sdp);
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for (i = 0; i < n_streams; i++) {
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@ -96,6 +96,7 @@ struct _GstRTSPStream {
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/* our udp sink back to the server */
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GstElement *udpsink;
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GstPad *rtcppad;
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/* state */
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gint pt;
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@ -142,6 +143,7 @@ struct _GstRTSPSrc {
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gchar *content_base;
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RTSPLowerTrans cur_protocols;
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gboolean tried_url_auth;
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gchar *addr;
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/* supported methods */
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gint methods;
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@ -47,18 +47,43 @@
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G_BEGIN_DECLS
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/**
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* RTSPTransMode:
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* @RTSP_TRANS_UNKNOWN: invalid tansport mode
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* @RTSP_TRANS_RTP: transfer RTP data
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* @RTSP_TRANS_RDT: transfer RDT (RealMedia) data
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*
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* The transfer mode to use.
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*/
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typedef enum {
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RTSP_TRANS_UNKNOWN = 0,
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RTSP_TRANS_RTP = (1 << 0),
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RTSP_TRANS_RDT = (1 << 1)
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} RTSPTransMode;
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/**
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* RTSPProfile:
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* @RTSP_PROFILE_UNKNOWN: invalid profile
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* @RTSP_PROFILE_AVP: the Audio/Visual profile
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* @RTSP_PROFILE_SAVP: the secure Audio/Visual profile
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*
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* The transfer profile to use.
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*/
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typedef enum {
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RTSP_PROFILE_UNKNOWN = 0,
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RTSP_PROFILE_AVP = (1 << 0),
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RTSP_PROFILE_SAVP = (1 << 1)
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} RTSPProfile;
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/**
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* RTSPLowerTrans:
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* @RTSP_LOWER_TRANS_UNKNOWN: invalid transport flag
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* @RTSP_LOWER_TRANS_UDP: stream data over UDP
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* @RTSP_LOWER_TRANS_UDP_MCAST: stream data over UDP multicast
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* @RTSP_LOWER_TRANS_TCP: stream data over TCP
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*
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* The different transport methods.
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*/
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typedef enum {
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RTSP_LOWER_TRANS_UNKNOWN = 0,
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RTSP_LOWER_TRANS_UDP = (1 << 0),
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@ -66,13 +91,26 @@ typedef enum {
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RTSP_LOWER_TRANS_TCP = (1 << 2)
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} RTSPLowerTrans;
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/**
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* RTSPRange:
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* @min: minimum value of the range
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* @max: maximum value of the range
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*
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* A type to specify a range.
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*/
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typedef struct
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{
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gint min;
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gint max;
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} RTSPRange;
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/**
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* RTSPTransport:
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*
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* A structure holding the RTSP transport values.
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*/
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typedef struct _RTSPTransport {
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/*< private >*/
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RTSPTransMode trans;
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RTSPProfile profile;
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RTSPLowerTrans lower_transport;
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