mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-24 17:20:36 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtspsrc: open on play and pause when not done yet

Browse source 
With the async state changes, it is possible that we need to open the stream
before play and pause.
Also make sure we remember a previous open failure so that we don't keep trying
again.
This commit is contained in:
Wim Taymans 2011-01-10 12:46:37 +01:00 committed by Mark Nauwelaerts
parent 6fe680934a
commit ddfcd8bbfd
2 changed files with 44 additions and 4 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

47
gst/rtsp/gstrtspsrc.c
View file

@ -3952,10 +3952,6 @@ gst_rtspsrc_reconnect (GstRTSPSrc * src, gboolean async)
/* we can try only TCP now */
src->cur_protocols = GST_RTSP_LOWER_TRANS_TCP;
/* pause to prepare for a restart */
if ((res = gst_rtspsrc_pause (src, FALSE, async)) < 0)
goto done;
/* close and cleanup our state */
if ((res = gst_rtspsrc_close (src, async)) < 0)
goto done;
@ -5744,11 +5740,13 @@ done:
no_sdp:
{
GST_WARNING_OBJECT (src, "can't get sdp");
src->open_error = TRUE;
goto done;
}
open_failed:
{
GST_WARNING_OBJECT (src, "can't setup streaming from sdp");
src->open_error = TRUE;
goto done;
}
}
@ -5986,6 +5984,30 @@ clear_rtp_base (GstRTSPSrc * src, GstRTSPStream * stream)
}
}
static GstRTSPResult
gst_rtspsrc_ensure_open (GstRTSPSrc * src, gboolean async)
{
GstRTSPResult res = GST_RTSP_OK;
if (src->state < GST_RTSP_STATE_READY) {
res = GST_RTSP_ERROR;
if (src->open_error) {
GST_DEBUG_OBJECT (src, "the stream was in error");
goto done;
}
if (async)
gst_rtspsrc_loop_start_cmd (src, CMD_OPEN);
if ((res = gst_rtspsrc_open (src, async)) < 0) {
GST_DEBUG_OBJECT (src, "failed to open stream");
goto done;
}
}
done:
return res;
}
static GstRTSPResult
gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
{
@ -5999,6 +6021,9 @@ gst_rtspsrc_play (GstRTSPSrc * src, GstSegment * segment, gboolean async)
GST_DEBUG_OBJECT (src, "PLAY...");
if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
goto open_failed;
if (!(src->methods & GST_RTSP_PLAY))
goto not_supported;
@ -6149,6 +6174,11 @@ done:
return res;
/* ERRORS */
open_failed:
{
GST_DEBUG_OBJECT (src, "failed to open stream");
goto done;
}
not_supported:
{
GST_DEBUG_OBJECT (src, "PLAY is not supported");
@ -6195,6 +6225,9 @@ gst_rtspsrc_pause (GstRTSPSrc * src, gboolean idle, gboolean async)
GST_DEBUG_OBJECT (src, "PAUSE...");
if ((res = gst_rtspsrc_ensure_open (src, async)) < 0)
goto open_failed;
if (!(src->methods & GST_RTSP_PAUSE))
goto not_supported;
@ -6261,6 +6294,11 @@ done:
return res;
/* ERRORS */
open_failed:
{
GST_DEBUG_OBJECT (src, "failed to open stream");
goto done;
}
not_supported:
{
GST_DEBUG_OBJECT (src, "PAUSE is not supported");
@ -6395,6 +6433,7 @@ gst_rtspsrc_thread (GstRTSPSrc * src)
src->cur_protocols = src->protocols;
/* first attempt, don't ignore timeouts */
src->ignore_timeout = FALSE;
src->open_error = FALSE;
ret = gst_rtspsrc_open (src, TRUE);
break;
case CMD_PLAY:

1
gst/rtsp/gstrtspsrc.h
View file

@ -182,6 +182,7 @@ struct _GstRTSPSrc {
gboolean ignore_timeout;
gboolean flushing;
gboolean waiting;
gboolean open_error;
/* mutex for protecting state changes */
GStaticRecMutex *state_rec_lock;