Scaletempo was missing an update of 'stop' in
new segment parameters when pushing it downstream,
which caused files to end earlier when rate < 1.
Fixes#599903
Based on patch by: Bastian Hecht <hechtb@gmail.com>
The scenario where you have a gap in a steady flow of packets of
say 10 seconds (500 packets of with duration of 20ms), the jitterbuffer
will idle up until it receives the first buffer after the gap, but will
then go on to produce 499 lost-events, to "cover up" the gap.
Now this is obviously wrong, since the last possible time for the earliest
lost-events to be played out has obviously expired, but the fact that
the jitterbuffer has a "length", represented with its own latency combined
with the total latency downstream, allows for covering up at least some
of this gap.
So in the case of the "length" being 200ms, while having received packet
500, the jitterbuffer should still create a timeout for packet 491, which
will have its time expire at 10,02 seconds, specially since it might
actually arrive in time! But obviously, waiting for packet 100, that had
its time expire at 2 seconds, (remembering that the current time is 10)
is useless...
The patch will create one "big" lost-event for the first 490 packets,
and then go on to create single ones if they can reach their
playout deadline.
See https://bugzilla.gnome.org/show_bug.cgi?id=667838
Ignore other commands when reconnecting, otherwise the loop function would pause
and the reconnection would not happen. Continue looping after doing a reconnect
so that we have a chance to actually read the new data.
The src pad caps always describe a single audio channel so only the
first position matters if deinterleave is configured to keep channel
positions in its src pads.
Make it possible to set the timeout after we went to the READY state by using
the timeout when checking the condition. This also makes it possible to set the
timeout with a higher granularity than seconds.
* gst/rtsp/gstrtspsrc.c (gst_rtspsrc_play): state change might be
happening in the application thread, so we don't change the state to
PLAYING in the gstrtspsrc thread unless it is safe.
A specific case is when chaning the state to NULL from the application
thread. This will synchronously try to stop the task (with the element
state lock acquired), but we will try a gst_element_set_state from
gstrtspsrc thread which will block on the element state lock causing a
deadlock.
https://bugzilla.gnome.org/show_bug.cgi?id=684312
Need to explicitly link against -lrt for clock_gettime(), which
we don't get in the libs any more, because core moved the
gmodule-no-export-2.0 bit into Requires.Private.
Not required for newer glibc, but for older ones, so check for that.
Shout2send only accepts webm format, not matroska, but due
to a bug in matroskamux, webmmux's source pad is also created
with the matroska source pad template as pad template, which
makes the link function think it can't link webmmux to shout2send.
Also add unit test.
https://bugzilla.gnome.org/show_bug.cgi?id=689336
Make a segment event when we send a new range header to a client (first PLAY
request or after a seek). Send the segment event in interleaved mode.
Clean the segment event on cleanup
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=688382
* gst/rtsp/gstrtspsrc.[ch]: added new "udp-reconnect" property. Before,
rtspsrc always tried to reconnect to the server when the RTSP
connection was closed by the server. This property lets the user
decide whether it wants rtspsrc to reconnect or not.
https://bugzilla.gnome.org/show_bug.cgi?id=683912
When the user configured a port-range, propose ports from this range
as the multicast ports. The server is free to ignore this request but if it
honours it, increment our ports so that we suggest the next port pair for the
next stream.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=639420
