This is to handle cases where upstream handles the fragmented streaming in TIME
segments and sends us data with gaps within fragments. This would happen when dealing
with trick-modes.

When upstream (push-based, TIME SEGMENT) wishes to send discontinuous samples,
it must obey the following rules:
* The buffer containing the [moof] must have a valid GST_BUFFER_OFFSET
* The buffers containing the first sample after a gap:
 * MUST start at the beginning of a sample,
 * MUST have the DISCONT flag set,
 * MUST have a valid GST_BUFFER_OFFSET relative to the beginning of the fragment.

https://bugzilla.gnome.org/show_bug.cgi?id=767354

gst_v4l2_clear_error() doesn't work like g_clear_error(), it
doesn't NULLify the pointer, so set freed debug string to NULL
so it doesn't get freed again if gst_v4l2_clear_error() is
called twice on the error.

CID 1362901

https://bugzilla.gnome.org/show_bug.cgi?id=768232

Commit 7873bede caused new caps
event per moof without consideration of duplication.

https://bugzilla.gnome.org/show_bug.cgi?id=768268

10 bytes was allocated for stream_format but size of "byte-stream" is
more. Use g_strdup() instead.

https://bugzilla.gnome.org/show_bug.cgi?id=753760

Otherwise set_flushing() will have undefined semantics and nowadays causes a
g_critical() to warn about that.

Update the blocksize depending on how much is obtained from a read
of the input stream. This avoids doing too many reads in small chunks
when larger amounts of data are available and also prevents using
a very large memory area to read a small chunk of data.

https://bugzilla.gnome.org/show_bug.cgi?id=767833

https://bugzilla.gnome.org/show_bug.cgi?id=767980

rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state

If we consider the RTSP state, what can happen is that it is PLAYING but the
element already asynchronously tried to PAUSE and it just did not happen yet.

We would then override this setting to PAUSED (while the element actually is
in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
to produce packets while the sinks are all PAUSED, piling up thousands of
packets in the rtpjitterbuffer and other elements and finally failing.

They are however supported by ffmpeg and apparently used out there.

https://bugzilla.gnome.org/show_bug.cgi?id=768006

https://bugzilla.gnome.org/show_bug.cgi?id=768006

This is partly automated using "make update" in docs/plugins, but also
required manual merge. Additionally, missing plugins and elements have
been added.

https://bugzilla.gnome.org/show_bug.cgi?id=767861

From ac2f647 to f363b32

Now we don't have to rely on a special value for the tile number.

https://bugzilla.gnome.org/show_bug.cgi?id=767817

gstrtpj2kpay.c:364:21: error: implicit truncation from 'int' to bitfield changes value from -1 to 65535

https://bugzilla.gnome.org/show_bug.cgi?id=767817

Need to call gst_check_drop_buffers() to release the buffers exchanged
during the test.

https://bugzilla.gnome.org/show_bug.cgi?id=766561

Flush the bus when cleaning up so pending messages are destroyed.

https://bugzilla.gnome.org/show_bug.cgi?id=766561

https://bugzilla.gnome.org/show_bug.cgi?id=766561

- Flush the bus so messages aren't leaked
- Fix pad leak

https://bugzilla.gnome.org/show_bug.cgi?id=766561

This is supposed to be either in the codec_data (avc stream format) or inside
the stream, and we extract it from there. It should not be set from a
property as it's stream specific.

https://bugzilla.gnome.org/show_bug.cgi?id=767789

The Session Data Protocol doesn't allow specifying a cipher for the
SRTCP, so it will use the SRTP one. In the "srtpenc" element the cipher
"aes-128-icm" is the default for SRTP and SRTCP, but if we want to have
an SRTCP with the "aes-256-icm" cipher then we also need to set the SRTP
cipher to "aes-256-icm", otherwise "aes-128-icm" will be used instead.

https://bugzilla.gnome.org/show_bug.cgi?id=767799

.. by registering the types from the plugin init function. This
seems to help, but we'll see if it's enough (might need similar
things elsewhere).

https://bugzilla.gnome.org/show_bug.cgi?id=693911
https://bugzilla.gnome.org/show_bug.cgi?id=674885

And the caps in the sink pad template already used variant (only).

There is no valid use case when this property is needed since the values
must be in either codec_data or buffer data.

https://bugzilla.gnome.org/show_bug.cgi?id=753760

Cosmetic change to read NALU type in gst_rtp_h265_pay_decode_nal() the
same way as in other places.

https://bugzilla.gnome.org/show_bug.cgi?id=753760

Documentation lacks '@' before each enum values and there was an extra
line after symbol section which confuses GTK-Doc parser.

https://bugzilla.gnome.org/show_bug.cgi?id=767788

https://bugzilla.gnome.org/show_bug.cgi?id=766025

Fixes sps/pps/vps insertion via the config-interval property.

https://bugzilla.gnome.org//show_bug.cgi?id=767680

Fixes parsing of "Thu May 11 15:57:46 2006 ".

https://bugzilla.gnome.org/show_bug.cgi?id=767496

This field is now required in the sink caps.

https://bugzilla.gnome.org/show_bug.cgi?id=766236

Seungha Yang authored Jun 09, 2016 and Nicolas Dufresne committed Jun 08, 2016

Fix unref assertion failure

https://bugzilla.gnome.org/show_bug.cgi?id=767424

With non-time segments, it now assumes that the arrival time of packets
is not relevant and that only the RTP timestamp matter and it produces
an output segment start at running time 0.

https://bugzilla.gnome.org/show_bug.cgi?id=766438