This new plugin helps in synchronizing multiple streams by clipping the beggining
of all streams until all streams are producing content.
This means, dropping content and rewriting timestamps to make all streams start at 0.

In JNI_OnLoad() we will already get the Java VM passed and could
just directly use that. gstreamer_android-1.0.c will now provide
this to us.

Reason for this is that apparently not all Android system are
providing the JNI functions to get the currently running Java VMs, so
we would fail to get. With this we will always be able to get the Java
VM on such systems.

Hardware codecs don't make sense for audio and are generally less reliable on
Android than the software codecs.

Someone else might have detached it before us, so make this just normal debug
output instead of a GST_ERROR()

We only need that if no Java VM is running yet, and all usual cases,
i.e. when calling GStreamer from an actual Android app, there will already
be a Java VM we can just use.

It seems like some phones come without that symbol, let's hope they come
with the other symbol but for now don't make a missing JNI_CreateJavaVM fatal.

Use QOS messages to update rendered and dropped frame stats. This is
the only accurate method. The old method didn't take max-lateness and
latency into account.

trivial patch to add proper ( while checking for if(G_UNLIKELY())

https://bugzilla.gnome.org/show_bug.cgi?id=751087

From 6015d26 to f74b2df

Getting the current viewport and modifying it relatively will produce an
interesting feedback loop during widget resizing. Over a few frames we
will gradually move the viewport a bit until it converged again, adding
unnecessary additional borders at the top and left.

CAIRO_FORMAT_ARGB32 is ARGB on big endian and BGRA on little endian.

After few iteration, this variable became the same as dts. It's not
the min as the name says, but the dts of the current buffer. Simply
remove and place with dts. Also move the debug trace to actually
print the signed version of the running-time dts.

videoaggregator: Remember if the last LATENCY query returned live or not and use the in the QoS messages

after e000a6f0, there is build error in bad plugins
this happens because, GST_CLOCK_STIME_IS_VALID () is being checked for pad_data
but it expects a GstClockTime parameter. Changing the check to 'dts'

https://bugzilla.gnome.org/show_bug.cgi?id=750961

It is not bound between 0 and 10000 anymore.

It is an unsigned integer so the upper bound is G_MAXUINT.

Luis de Bethencourt authored Oct 29, 2014 and Tim Müller committed Jun 13, 2015

Switch the increment of markersize from when it is used to when it is
returned from compute_resync_marker_size.

This also makes the CHECK_REMAINING in gst_mpeg4_parse_video_packet_header
check for the actually required number of bits now and not one too few.

https://bugzilla.gnome.org/show_bug.cgi?id=739345

This reverts commit 916b9543.

Clearly something else was intended, and it also makes
more sense to add the extra bit. The resync marker is
N zero bits plus a 1 bit, and the pattern/mask needs to
be run on N+1 bits too.

(Even after the rever the code doesn't do that of course, so
it still needs to be fixed differently.)

https://bugzilla.gnome.org/show_bug.cgi?id=739345

Headers were broken on older Android versions, apparently.

https://bugzilla.gnome.org/show_bug.cgi?id=744459

This allows us to signal what kind of audio we are expecting to record,
which should tell the system to apply filters (such as echo
cancellation, noise suppression, etc.) if required.

This was fixed by https://bugzilla.gnome.org/show_bug.cgi?id=749258
in basesink, and is not necessary to duplicate here anymore.

https://bugzilla.gnome.org/show_bug.cgi?id=750881