Corrected some comments in gstmpdparser.h file.
Moved gst_mpd_client_get_adaptation_sets function to be grouped with
other functions from AdaptationSet group
https://bugzilla.gnome.org/show_bug.cgi?id=751149
The gst_mpdparser_get_rep_idx_with_max_bandwidth function assumes
representations are ordered by bandwidth and incorrectly returns the
first one when wanting the one with minimum bandwidth.
Corrected gst_mpdparser_get_rep_idx_with_max_bandwidth function to get the
correct representation in case max_bandwidth parameter is 0.
https://bugzilla.gnome.org/show_bug.cgi?id=751153
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.
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.
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.
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.
after e000a6f0a4, 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
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 916b954315.
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
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.
We now know that pool caching can cause renegotiation issues
when an element in the pipeline change from passthrough to not
passthrough. As it's not needed, don't cache existing pools.
https://bugzilla.gnome.org/show_bug.cgi?id=748344
The segment should start at first PTS, and the vairable name lower_pts
state so correctly. Though we where using the first DTS instead. This
could lead to small desynchronization of video stream.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
Use the saved DTS, make it signed and pass that to the stream muxer. This
preserves the running time sign. All usage of -1 as invalid TS are now
replaced with G_MININT64. Negative values will be seen as wrap-around
point, but the delta between PTS and DTS will remain correct. Demuxers
don't care about absolute values, they only cares about deltas.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
There was code to detect backward dts, but the marker min_dts
was never set. Setting it enable this feature that prevents
potential integer overflow when generating TS.
https://bugzilla.gnome.org/show_bug.cgi?id=740575
Added a check for a_node->ns before accessing a_node->ns->href in
gst_mpdparser_get_xml_node_namespace. This could happen if the xml
is missing the default namespace.
https://bugzilla.gnome.org/show_bug.cgi?id=750866
An example app that takes video URIs as command line arguments and switches
between them seamlessly one after the other using compositor and audiomixer.
Both audio-video and video-only media files are valid inputs, but mixing files
of both types in a single invocation is cumbersome to support, and hence does
not work. The example attempts to keep the audio stream moving along perfectly,
and duplicates video frames where necessary to cover gaps in the video
timestamps using the 'ignore-eos' videoaggregator pad property.
Ensuring seamless (and mostly-glitch-free) switching is harder than it sounds,
and hence the example contains plenty of pad probes and running time
calculations to make things work.
The GPtrArray play_queue contains items that are being played back, have been
prepared for playback, and will be played back in the future. The queue itself
is mutable besides the first two items (playing and prepared). The item that has
been prepared should not be edited or removed since it has been prepared in
advance to be activated immediately on the current item's EOS.
The example also has support for switching to the next item in the queue
prematurely; see the --switch-after/-s flag to the application.
Note: the output video is hard-coded at 1280x720, and input video is scaled as
needed to fit this size. Set OUTPUT_VIDEO_WIDTH/HEIGHT to change this.
https://bugzilla.gnome.org/show_bug.cgi?id=748947
When the 'ignore-eos' property is set on a pad, compositor will keep resending
the last buffer on the pad till the pad is unlinked. We count the buffers
received on appsink, and if it's more than the buffers sent by videotestsrc, the
test passes.
