We detect a container correctly now so we need to revert the weird
check there was before.
Use gst_rtspsrc_stream_push_event() to push the caps event on the
right pad.
See https://bugzilla.gnome.org/show_bug.cgi?id=739391
Keep global and stream tags separately and parse the udta node
that can be found under the trak atom. The udta will contain
stream specific tags and will be pushed as such
https://bugzilla.gnome.org/show_bug.cgi?id=692473
Tags received via events, when marked as stream tags, will
be stored on that stream's trak atom instead of being stored
in the main tags atom. This allows the resulting file to have
global and stream tags stored.
https://bugzilla.gnome.org/show_bug.cgi?id=692473
Refactor the functions that were bound to the 'moov' atom to
directly pass the desired 'udta' that should receive the tags.
This allows the tags to be written to 'udta' at the 'moov' or
the 'trak' level, creating tags that are for the container or
for a stream only.
https://bugzilla.gnome.org/show_bug.cgi?id=692473
Snap to the end of the file when seeking past the end in reverse mode,
and also fix GST_SEEK_TYPE_END and GST_SEEK_TYPE_NONE handling
for the stop position by always seeking on a segment in stream time
This will be emitted whenever an RTCP packet is received. Different to
on-feedback-rtcp, this signal gets every complete RTCP packet and not
just the individual feedback packets.
For fragmented streams with extra data at the end of the mdat
qtdemux was not dropping those bytes and would try to use
that extra data as the beginning of a new atom, causing the
stream to fail.
https://bugzilla.gnome.org/show_bug.cgi?id=743407
It had no effect since quite some time and also is not needed in general,
especially not to switch between immediate feedback mode and early feedback
mode. The latest understanding of the RFC is that from the endpoint point of
view, both modes are exactly the same. RTCP is only allowed to use the
bandwidth as given by the RFC constraints, as such it is only ever possible
to schedule a RTCP packet early but it's against the RFC to schedule more RTCP
packets.
The difference between immediate feedback mode and early feedback mode is that
the former guarantees that an RTCP packet can be sent for every event
"immediately", which means that the bandwidth calculations from the RFC have
resulted in an RTCP scheduling interval that is small enough. Early feedback
mode on the other hand means that we can schedule some packets early to make
that happen, but it's not guaranteed at all that it's possible to schedule
an RTCP packet per event (i.e. they need to be accumulated or dropped).
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
