Improve buffer validation by making sure each memory are the right
one and that each memory is writable. This fixes tearing issues in
case downstream uses gst_buffer_make_writable() or other type
of GstBuffer copy where memory are only reffed.
https://bugzilla.gnome.org/show_bug.cgi?id=739754
Right now in parse logic the signature is checked every time the parse function
is called, and the whole data is the scanned each and every time, even though the
data is scanned in the previous instance. Changing the logic such that, we skip
the bytes which are already scanned in the previous instances of parse. This
helps in avoiding multiple scan of already scanned data/signature.
https://bugzilla.gnome.org/show_bug.cgi?id=737708
Rather than try and guess interlace support as part of checking supported
sizes, look for interlace support specifically in its own function.
As a cleanup, use V4L2_FIELD_ANY when probing sizes, which should result in
the driver doing the right thing.
With my capture setup, this gets me the following sample caps:
For 1080i resolution:
video/x-raw, format=(string)YUY2, width=(int)1920, height=(int)1080, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)interleaved, framerate=(fraction){ 25/1, 30/1 }
For 720p resolution:
video/x-raw, format=(string)YUY2, width=(int)1280, height=(int)720, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, framerate=(fraction){ 50/1, 60/1 }
For 576i/p resolution (both possible at the point of query):
video/x-raw, format=(string)YUY2, width=(int)720, height=(int)576, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string){ progressive, interleaved }, framerate=(fraction){ 25/1, 50/1 }
This, in turn, makes 576i work correctly; with the old code,
the caps would be interlace-mode=progressive for interlaced video.
https://bugzilla.gnome.org/show_bug.cgi?id=726194
Both Firefox and Chrome uses VP8 as the encoding in their SDP.
Adding this now defacto standard name removes the need for special
case in SDP parsing code.
https://bugzilla.gnome.org/show_bug.cgi?id=737810
Add fixed payload type for mp2t to template caps as well, so
our output caps match the advertised default pt. Fixes a
regression from 1.2.
There's still something wrong with caps negotiation though,
rtpmp2tpay payload=96 ! fakesink will not output caps with
payload=96.
On streamon failure, the queued buffer is not released from the
bufferpool class point of view because it is queued to the driver and
the flush logic is not performed since we are not in streaming state.
It causes the v4l2 bufferpool to always return that stop method failed
and to leak v4l2 objects and buffers.
This commit solve this by performing the flush logic in error case, ie
flushing the allocator and restoring queued buffer state to non-queued.
https://bugzilla.gnome.org/show_bug.cgi?id=738102
This will prevent deadlocks, but will also properly flush the pool and allocator
when going to READY state. It should also fix issues reported on mailing list
when seeking is performed.
https://bugzilla.gnome.org/show_bug.cgi?id=738152
Some of the RTP unit tests are very flaky and will
fail more often with the CPU maxed out fully. Those
tests need to be fixed in any case though, they also
fail on slower machines and also occasionally with
normal 'make check'.
Fixes crash in audiotestsrc because of an unsupported format
getting negotiated on big-endian systems with
audiotestsrc ! interleave ! audioconvert ! wavenc
We need a mechanism in PulseAudio to allow running code outside the
mainloop lock. Then we'd be able to post to the bus (taking the
GST_OBJECT_LOCK), without worrying about locking order with the mainloop
lock, which is the current cause of deadlocks while trying to post the
stream status messages.
https://bugzilla.gnome.org/show_bug.cgi?id=736071
When the RTT and jitter are very low (such as on a local network), the
calculated retransmission timeout is very small. Set some sensible lower
boundary to the timeout by adding a new property. We use the packet
spacing as a lower boundary by default.
In early retransmission we are allowed to schedule 1 regular RTCP packet
at an earlier time. When we do that, we need to set allow_early to FALSE
and ignore/drop (or merge) all future requests for early transmission.
We now first check if we can schedule an early RTCP and if we can,
actually prepare the data for the next RTCP interval.
After we send the next regular RTCP after the early RTCP, we set
allow_early to TRUE again to allow more early requests.
Remove the condition for the immediate feedback for now.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
Add a need-resync state, this is when we need to try to lock on to a
time/RTPtime pair.
Always check the RTP timestamps and if they go backwards, mark ourselves
as need-resync.
Only resync when need-resync is TRUE and we have a valid time. Otherwise
we keep the old values. This avoids locking on to an invalid time and
causing us to timestamp everything with -1.
Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
rtpmux behaves like a funnel in that it forwards whatever upstream is
sending buffers. So setting proxy caps doesn't make sense as the
upstream don't have to have compatible caps, thus resulting in an empty
caps set as a result of a caps query. Instead set fixed caps just
as funnel does.
https://bugzilla.gnome.org/show_bug.cgi?id=738722
left, right, top, bottom can be set from range of -2147483648 to 2147483647
when i launch the videobox element with that values, it gives a critical error
(gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
This happens because min cannot be equal to max.
https://bugzilla.gnome.org/show_bug.cgi?id=738838
