Otherwise setting an URI after creation will already set the state
to READY/buffering and disallow setting the configuration.
See https://github.com/servo/servo/issues/22010
This reverts commit 73ebdb888e.
This isn't needed and it breaks srtpenc ! srtpdec, specifying the
roll-over counter manually is an advanced feature.
Also revert "srtp: Add "roc" caps field to the gst-launch example"
This reverts commit 67ae35813b.
https://bugzilla.gnome.org/show_bug.cgi?id=765079
ext/sctp/ext@sctp@@gstsctp@sha/sctpassociation.c.obj: In function `receive_cb':
/var/lib/jenkins/workspace/cerbero-cross-mingw32/workdir/sources/windows_x86/gst-plugins-bad-1.0-1.15.0.1/_builddir/../ext/sctp/sctpassociation.c:692: undefined reference to `_imp__ntohl@4'
Expanded to support image format to YV12/I422/I444. It's related to the
color bit-depth and profile of the codec. It can make configuring
appropriate profile according to bit-depth and format.
https://bugzilla.gnome.org/show_bug.cgi?id=791674
There's a race condition when is-live is set to true and the shmsrc
element releases the pipe in the transition from PLAYING to PAUSED.
To avoid it this change ensures that _create method takes the pipe
and increases the use_count in one operation protected by object lock.
Also perform apropriate protections when releasing the pipe.
https://bugzilla.gnome.org/show_bug.cgi?id=797203
With prenegotiated channels, the data-channel protocol is not used and
instead the channel's negotiation is intended to be performed out of band in
some application-specific manner.
Comes with test!
This means that we will reject all operations before we've transitioned
into READY.
This also fixes the tests using the default GMainContext in the NULL
state instead of the webrtcbin internal GMainContext and thread. Also
removes a potential ordering race where on the element transitioning to
READY, an operations could have been queued on two different threads and
removing a guarentee on operation ordering.
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.
