mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-01 20:30:41 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtspsrc: Also handle redirect on PLAY

Browse source 
https://bugzilla.gnome.org/show_bug.cgi?id=772610
This commit is contained in:
Matt Staples 2016-10-21 17:31:00 +00:00 committed by Sebastian Dröge
parent 5cdf66d5d2
commit cd71e3a8e8
1 changed files with 33 additions and 24 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

57
gst/rtsp/gstrtspsrc.c
View file

@ -343,13 +343,13 @@ typedef struct
} PtMapItem;
/* commands we send to out loop to notify it of events */
#define CMD_OPEN (1 << 0)
#define CMD_PLAY (1 << 1)
#define CMD_PAUSE (1 << 2)
#define CMD_CLOSE (1 << 3)
#define CMD_WAIT (1 << 4)
#define CMD_RECONNECT (1 << 5)
#define CMD_LOOP (1 << 6)
#define CMD_OPEN (1 << 0)
#define CMD_PLAY (1 << 1)
#define CMD_PAUSE (1 << 2)
#define CMD_CLOSE (1 << 3)
#define CMD_WAIT (1 << 4)
#define CMD_RECONNECT (1 << 5)
#define CMD_LOOP (1 << 6)
/* mask for all commands */
#define CMD_ALL ((CMD_LOOP << 1) - 1)
@ -4881,30 +4881,26 @@ gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
if (!restart)
goto done;
/* unless redirect, try tcp */
if (!src->need_redirect)
src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
/* we can try only TCP now */
src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
/* close and cleanup our state */
if ((res = gst_rtspsrc_close (src, async, FALSE)) < 0)
goto done;
/* unless redirect, see if we have TCP left to try. Also don't
* try TCP when we were configured with an SDP. */
if (!src->need_redirect && (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP)
|| src->from_sdp))
/* see if we have TCP left to try. Also don't try TCP when we were configured
* with an SDP. */
if (!(src->protocols & GST_RTSP_LOWER_TRANS_TCP) || src->from_sdp)
goto no_protocols;
if (!src->need_redirect) {
/* We post a warning message now to inform the user
* that nothing happened. It's most likely a firewall thing. */
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. Retrying using a tcp connection.",
gst_guint64_to_gdouble (src->udp_timeout) / 1000000.0));
}
/* We post a warning message now to inform the user
* that nothing happened. It's most likely a firewall thing. */
GST_ELEMENT_WARNING (src, RESOURCE, READ, (NULL),
("Could not receive any UDP packets for %.4f seconds, maybe your "
"firewall is blocking it. Retrying using a tcp connection.",
gst_guint64_to_gdouble (src->udp_timeout) / 1000000.0));
/* unless redirect, open new connection using tcp */
/* open new connection using tcp */
if (gst_rtspsrc_open (src, async) < 0)
goto open_failed;
@ -6770,7 +6766,7 @@ restart:
NULL)) < 0)
goto send_error;
/* we only perform redirect for the describe, currently */
/* we only perform redirect for describe and play, currently */
if (src->need_redirect) {
/* close connection, we don't have to send a TEARDOWN yet, ignore the
* result. */
@ -7256,6 +7252,7 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
GST_DEBUG_OBJECT (src, "PLAY...");
restart:
if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
goto open_failed;
@ -7328,6 +7325,18 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
if ((res = gst_rtspsrc_send (src, conn, &request, &response, NULL)) < 0)
goto send_error;
if (src->need_redirect) {
GST_DEBUG_OBJECT (src,
"redirect: tearing down and restarting with new url");
/* teardown and restart with new url */
gst_rtspsrc_close (src, TRUE, FALSE);
/* reset protocols to force re-negotiation with redirected url */
src->cur_protocols = src->protocols;
gst_rtsp_message_unset (&request);
gst_rtsp_message_unset (&response);
goto restart;
}
/* seek may have silently failed as it is not supported */
if (!(src->methods & GST_RTSP_PLAY)) {
GST_DEBUG_OBJECT (src, "PLAY Range not supported; re-enable PLAY");