It might be possible that if we set webrtcbin to the NULL state some
tasks (idle sources) are still executed and they might even freeze. The freeze
is caused because the webrtcbin tasks don't hold a reference to webrtcbin and
if it's last unref inside the idle source itself this will not allow the main
loop to finish because the main loop is waiting on the idle source to finish.
We now start and stop webrtcbin thread when changing states. This will allow
the idle sources to finish properly.
https://bugzilla.gnome.org/show_bug.cgi?id=797251
If the first audio buffer to be dropped started right between two video
buffers (after the end of the first but before the start of the second,
as is often the case with N/1001 video frame rates), we would miss
sending the dropping=true message.
https://bugzilla.gnome.org/show_bug.cgi?id=797248
tsdemux expects a custom descriptor (GST_MTS_DESC_AC3_AUDIO_STREAM)
to detect a stream as AC3 and not EAC3.
Note that tsdemux expects this descriptor because mpegtsmux writes
a stream with a HDMV registration descriptor.
Fixes:
gst-launch-1.0 -v audiotestsrc ! avenc_ac3 ! ac3parse ! mpegtsmux ! \
tsdemux ! ac3parse ! avdec_ac3 ! audioconvert ! autoaudiosink
https://bugzilla.gnome.org/show_bug.cgi?id=797220
For each lib we build export its own API in headers when we're
building it, otherwise import the API from the headers.
This fixes linker warnings on Windows when building with MSVC.
The problem was that we had defined all GST_*_API decorators
unconditionally to GST_EXPORT. This was intentional and only
supposed to be temporary, but caused linker warnings because
we tell the linker that we want to export all symbols even
those from externall DLLs, and when the linker notices that
they were in external DLLS and not present locally it warns.
What we need to do when building each library is: export
the library's own symbols and import all other symbols. To
this end we define e.g. BUILDING_GST_FOO and then we define
the GST_FOO_API decorator either to export or to import
symbols depending on whether BUILDING_GST_FOO is set or not.
That way external users of each library API automatically
get the import.
While we're at it, add new GST_API_EXPORT in config.h and use
that for GST_*_API decorators instead of GST_EXPORT.
The right export define depends on the toolchain and whether
we're using -fvisibility=hidden or not, so it's better to set it
to the right thing directly than hard-coding a compiler whitelist
in the public header.
We put the export define into config.h instead of passing it via the
command line to the compiler because it might contain spaces and brackets
and in the autotools scenario we'd have to pass that through multiple
layers of plumbing and Makefile/shell escaping and we're just not going
to be *that* lucky.
The export define is only used if we're compiling our lib, not by external
users of the lib headers, so it's not a problem to put it into config.h
Also, this means all .c files of libs need to include config.h
to get the export marker defined, so fix up a few that didn't
include config.h.
This commit depends on a common submodule commit that makes gst-glib-gen.mak
add an #include "config.h" to generated enum/marshal .c files for the
autotools build.
https://bugzilla.gnome.org/show_bug.cgi?id=797185
Previously it was dispatched before the last video buffer, and audio
buffers would follow afterwards. It's misleading to send the
dropping=true message before both streams have really stopped, it can
lead to races when someone is e.g. waiting for that message to send EOS.
Also added some debug output.
https://bugzilla.gnome.org/show_bug.cgi?id=797145
Fixes a race where the task could attempt to set
stream-start/caps/segment before the pad was active and would be
dropped resulting in a 'data-flow before stream-start' warning.
It is possible and often desirable to pass multiple ICE relays
to libnice agents, the "turn-server" property, while convenient
to use from the command line, does not allow that.
This adds a new action signal, "add-turn-server" to address that.
https://bugzilla.gnome.org/show_bug.cgi?id=797012
../sys/decklink/gstdecklinkvideosink.cpp:1006:11: error: ‘GstDecklinkVideoSink {aka struct _GstDecklinkVideoSink}’ has no member named ‘scheduled_stop_time’
self->scheduled_stop_time = start_time;
^
Decklink sometimes does not notify us through the callback that it has
stopped scheduled playback either because it was uncleanly shutdown
without an explicit stop or for unknown other reasons.
Wait on the cond for a short amount of time before checking if scheduled
playback has stopped without notification.
https://bugzilla.gnome.org/show_bug.cgi?id=797130
This is part of a much larger goal to always keep the frames we schedule to
decklink be always increasing. This also allows us to avoid using both the
sync and async frame display functions which aren't recomended to be used
together.
If the output timestatmsp is not always increasing decklink seems to hold
onto the latest frame and may cause a flash in the output if the played
sequence has a framerate less than the video output.
Scenario is play for N seconds, pause, flushing seek to some other position,
play again. Each of the play sequences would normally start at 0 with
the decklink time. As a result, the latest frame from the previous sequence
is kept alive waiting for it's timestamp to pass before either dropping
(if a subsequent frame in the new sequence overrides it) or displayed
causing the out of place frame to be displayed.
This is also supported by the debug logs from the decklink video sink
element where a ScheduledFrameCompleted() callback would not occur for
the frame until the above had happened.
It was timing related as to whether the frame was displayed based
on the decklink refresh cycle (which seems to be 16ms here),
when the frame was scheduled by the sink and the difference between
the 'time since vblank' of the two play requests (and thus start times
of scheduled playback).
https://bugzilla.gnome.org/show_bug.cgi?id=797130
Direct applying the commit 7bb6443. This could fix also unexpected
nal dropping when nonzero "config-interval" is set.
(e.g., gst-launch-1.0 videotestsrc ! x265enc key-int-max=30 !
h265parse config-interval=30 ! avdec_h265 ! videoconvert ! autovideosink)
Similar to the h264parse, have_{vps,sps,pps} variables will be used
for deciding on when to submit updated caps or not, and rather mean
"have new SPS/PPS to be submitted?"
See also https://bugzilla.gnome.org/show_bug.cgi?id=732203https://bugzilla.gnome.org/show_bug.cgi?id=754124
