Don't reset the time in flush-stop. Live sources can do this flush in the
playing state and so the pipeline will never have a chance to update the
base_time of the elements, which only happens when going from paused to
playing.
Always send the flushing events to the udp elements now that basesrc supports
this. This makes sure a segment event is sent correctly after a flush.
Keep track of the currently executing command and make it possible to specify
what command you want to cancel when starting a new async command.
See https://bugzilla.gnome.org/show_bug.cgi?id=677905
These should all be int instead of long, to avoid bugs
when passing these as varargs with g_object_set(), and
there was no reason to use long in the first place here.
Fixes FIXME.
They should take the filter caps into account and always return
the template caps appended to the actual caps. Otherwise the
parsers stop to accept unparsed streams where upstream does not
know about channels, rate, etc.
Fixes bug #677401.
From RFC 2250
2. Encapsulation of MPEG System and Transport Streams
...
For MPEG2 Transport Streams the RTP payload will contain an integral
number of MPEG transport packets. To avoid end system
inefficiencies, data from multiple small MTS packets (normally fixed
in size at 188 bytes) are aggregated into a single RTP packet. The
number of transport packets contained is computed by dividing RTP
payload length by the length of an MTS packet (188).
....
Since it needs to contain "an integral number of MPEG transport packets", a
simple fix is to check that's the case, and strip off any leftover data.
Fixes#676799
Conflicts:
gst/rtp/gstrtpmp2tdepay.c
demux->common.segment is populated during seek handling with the target
start/stop positions. Don't override them when sending out a NEWSEGMENT.
Conflicts:
gst/matroska/matroska-demux.c
If there are no profile restrictions downstream, return caps with
profile=constrained-baseline in the first structure and append
unrestricted caps as the last structure.
Fixes bug #672019
Setting GST_SEEK_FLAG_SKIP when sending a seek event in rtspsrc should
set the "Scale" field in the rtsp PLAY header.
Because the boolean "src->skip" is set after the call, "Speed" instead
of "Scale" is always set. Move the assignment before issuing the _play
request.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=676618
This fixes a not-negotiated error at least on mov files with
twos audio with two channels and video dvcp. As playbin and gst-launch
sample coming from the qtdemux.c file uses audioconvert and the latter
require format interleaved.
https://bugzilla.gnome.org/show_bug.cgi?id=675326
When all pads go to EOS immediately, we are not negotiated and our collected
function is called (without any available data). Handle this case gracefully.
Conflicts:
gst/interleave/interleave.c
