Use gst_gl_sized_gl_format_from_gl_format_type to get the format passed to
CVOpenGLESTextureCacheCreateTextureFromImage. Before this change extracting the
second texture from the pixel buffer was failing on ios 9.1.
It's for the upstream element driving the pipeline to
handle seeks and send flush events or not, filters
should not interfere here, otherwise downstream pads
could be flushing before upstream pads are flushing,
which can result in GST_FLOW_ERROR being sent instead
of GST_FLOW_FLUSHING when trying to forward sticky
events at just the wrong moment.
It is up to the element handling the seek to send flush events
downstream, otherwise we end up with a situation where upstream
would get unexpected GST_FLOW_FLUSHING
The Onvif Streaming Specification specifies that the NTP timestamps
in the Onvif extension header indicaes the absolute UTC time associated
with the access unit. But by using running time we can not achieve that,
since a frame's running time depends on the played interval, whether a
non-flushing is done, etc. Instead we have to use the stream time.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
It is now possible to update the currently used ntp-offset with a
custom serialized downstream event. The element will read the ntp-offset
property when doing the state transition from READY to PAUSED and
use that offset until it receives a "GstNtpOffset" event, which also
has a "ntp-offset" attribute in that it's structure. In case the
property is not set and no event has been received, the element will
guess the npt-offset with help of the clock. If no clock can be
retrieved, the element will error out and stop the data flow.
The same event is also used for updating the D/E-bits in the RTP
extension header. The discont flag in a buffer can be set whenver a
live/network source looses a frame, but that is not the type of
discontinuity that the onvif extension header should reflect. The
header is mainly used for playback of a track concept, in which
gaps can be present, and it's those kind of gaps that should be
highlighted with the D- and E-bits.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
If a buffer or a buffer list is cached, no events serialized with the
data stream should get through. The cached buffers and events should
be purged when we stop flushing.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
Split the unit tests for rtponviftimestamp and rtponvifparse
elements in separate files.
Setup and cleanup the element and pads in fixures. Make the tests work
with CK_FORK=no as well, by cleaning up the 'buffers' list when needed.
Make unit tests work when run in valgrind by unreffing all buffers,
and by not allocating any payload in RTP buffers. Since we're not
doing anything with the payload part, but we're memcmp-aring the
complete buffer memory, valgrind complained about non-initialized
memory being used.
https://bugzilla.gnome.org/show_bug.cgi?id=757688
Bitrate estimation is now handled through a queue2 element added after
the source element used to download fragments.
Original hlsdemux patch by Duncan Palmer <dpalmer@digisoft.tv>
https://bugzilla.gnome.org/show_bug.cgi?id=733959
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_FORMAT.
Plus it creates more readable values in the logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
We always require the channel-mapping-field. If it's 0 we require nothing
else, otherwise we need channels, stream-count and coupled count to be
available.
oggdemux is outputting the meta now, and only outputs if it should really
apply to the current buffer. Previously we would skip N samples also if we
started the decoder in the middle of the stream.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
It is doing the wrong thing because of the Opus pre-skip: while the timestamps
are shifted by the pre-skip, the granule positions are not shifted.
oggmux is doing the right thing here already.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
The first frame has lookahead less samples, the last frame might have some
padding or we might have to encode another frame of silence to get all our
input into the encoded data.
This is because of a) the lookahead at the beginning of the encoding, which
shifts all data by that amount of samples and b) the padding needed to fill
the very last frame completely.
Ideally we would use LPC to calculate something better than silence for the
padding to make the encoding as smooth as possible.
With this we get exactly the same amount of samples again in an
opusenc ! opusdec pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=757153
No need to use G_GINT64_FORMAT for potentially negative values of
GstClockTimeDiff. Since 1.6 these can be handled with GST_STIME_ARGS.
Plus it creates more readable values in the logs.
https://bugzilla.gnome.org/show_bug.cgi?id=757480
Waylandsink needs exception code in gst_wayland_sink_set_window_handle().
After making sink->window, User can call
gst_wayland_sink_set_window_handle(). It is the user's fault, but
Waylandsink needs to handle the exception, if not then sink->window is
changed and rendering fails.
https://bugzilla.gnome.org/show_bug.cgi?id=747482
Waylandsink needs exception code in gst_wayland_sink_set_context(). After
calling gst_wayland_sink_set_context(), below code is set.
GST_ELEMENT_CLASS (parent_class)->set_context (element, context); but, If
user can call onemore. It is user's fault. but waylandsink need to
exception.
https://bugzilla.gnome.org/show_bug.cgi?id=747482
This code will never be called as max>=min in all cases. If the upstream
latency query returned min>max, the function already returned and all
values that are added to those have max>= min.
