This indicates with a boolean return value if scheduling a new RTCP packet
within the requested delay was possible. Otherwise it behaves exactly like
send-rtcp. The only reason for adding a new signal is ABI compatibility.
If we can not create probe stream in query_getcaps function, it will appear
memory leakage from format info.
The following patch prevent memory leakage in pulsesink.
https://bugzilla.gnome.org/show_bug.cgi?id=743178
No matter if gst_matroska_read_common_parse_index_cuetrack () returns that the
flow is OK or not, the check there will be a break from the switch. Removing the
check since the outcome is the same.
CID #1265762
Replace the sink_query with new getcaps() virtual and use the proxy
helper with the probed caps. This allow upstream element taking decision
base on what is supported downstream.
The hack causes deadlocks and other interesting problems and it really
can only be fixed properly inside GLib. We will include a patch for
GLib in our builds for now that handles this, and hopefully at some
point GLib will also merge a proper solution.
A proper solution would first require to refactor the polling in
GMainContext to only provide a single fd, e.g. via epoll/kqueue
or a thread like the one added by our patch. Then this single
fd could be retrieved from the GMainContext and directly integrated
into a NSRunLoop.
https://bugzilla.gnome.org/show_bug.cgi?id=741450https://bugzilla.gnome.org/show_bug.cgi?id=704374
v4l2loopback driver has a this nasty bug that if the queue is larger
then 2 buffers, it returns random index on dqbuf. So far we assumed
that the index was always right, which would lead to memory being
unref twice, and eventually crash.
As the buffer array is fixed size and small, it's safer to simply
use this static size to cleanup the buffers. This is also more
consistent with the rest. The associated method is no longer
required and can be dropped.
This partly revert to the old 1.2 behavior. Instead of keeping a
reference to the output buffer queued, we simply release them but
don't forward it to GstBufferPool. This way, the buffer pool don't
need to be flushed to be stopped.
https://bugzilla.gnome.org/show_bug.cgi?id=742074
Failing streamoff prevents allocator from being disposed hence
lead to device FD leak. There is no known cases where streamoff
may fails for which we'd still be streaming. streamoff is known
to fail when a device is being unplugged (in which case errno
19/ENODEV is set).
https://bugzilla.gnome.org/show_bug.cgi?id=732734
Bringing back the check removed in the previous commit but have that check be a
g_assert. Changing the function to static void since return can never be False,
because audio format will never be unkown.
func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
The conditional checking if func_index is >= 0 and < 8 will always be true.
Removing it.
CID 1226442
We (currently?) can't really handle gaps between RTP packets if they're not
properly timestamped. The current code would go into calculations with
GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
better to error out cleanly instead.
We set to PLAYING after we have configured the caps, otherwise we
might end up calling request_key (with SRTP) while caps are still
being configured, ending in a crash.
https://bugzilla.gnome.org/show_bug.cgi?id=740505
They should always be built, while the speex elements are not.
Need to check for a smaller number of buffers then (7->4) because
speexenc will add 3 header buffers while alawenc will just output
as many buffers as it receives as input.
https://bugzilla.gnome.org/show_bug.cgi?id=742098
basesrc assumes that we don't return a buffer if
something else than OK is returned. It will just
leak any buffer we might accidentially provide
here.
This can potentially happen during flushing.
Maybe fixes https://bugzilla.gnome.org/show_bug.cgi?id=741993
Actually copy the codec data instead of copying nothing
and then bombing out because there's no data.
Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink
https://bugzilla.gnome.org/show_bug.cgi?id=741783
Apparently linphone sends an invalid RTP packet as very
first packet. We want to ignore that instead of erroring
out (same for any other invalid packets really).
https://bugzilla.gnome.org/show_bug.cgi?id=741398
Set positioning-mode=pixels-absolute to allow positioning with
absolute coordinates, meaning negative x/y offsets will be
interpreted as being to the left/above the video frame instead
of being interpreted as relative to the right/bottom edge of
the video frame (which is a silly default, but that's how it is).
This means we can nicely slide images into and out of the frame,
see gdkpixbufoverlay-test.
https://bugzilla.gnome.org/show_bug.cgi?id=739566
