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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!
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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.
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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
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Alpha in GStreamer is unpremultiplied
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Flattens the overlay compositions into the stream before the mixer will
mix them.
https://bugzilla.gnome.org/show_bug.cgi?id=759867
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GST_TYPE_GL_STEREO_DOWNMIX_MODE_TYPE
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Also update the property description and range with aggregator's values.
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Mostly follows the W3C specification
https://www.w3.org/TR/webrtc/#peer-to-peer-data-api
With contributions from:
Mathieu Duponchelle <mathieu@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=794351
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Can occur if no connection is actually made and thus no connection
thread is created.
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They can be set later
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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.
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- Add meson build definitions
- Add necessary API decorators
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https://bugzilla.gnome.org/show_bug.cgi?id=744863
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g_resolver_lookup_by_name is a blocking call, and should not
be run when the user sets or adds a turn-server.
https://bugzilla.gnome.org/show_bug.cgi?id=797012
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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
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cl : Command line error D8021 : invalid numeric argument '/Wno-missing-include-dirs'
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We now have options for all plugins, so we will just disable these in
the cerbero recipe instead. These require external deps, so they won't
affect gst-build either.
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Although RTMP_ConnectStream() was failed, librtmp's internal memory
is not freed by RTMP_ConnectStream(), so RTMP_Close() should be called
before RTMP_Free()
https://bugzilla.gnome.org/show_bug.cgi?id=797058
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Worst case it will be empty. This fixes a crash when the base class
calls data_received() when the stream is neither is_isobmff or
has_isoff_ondemand_profile.
https://bugzilla.gnome.org/show_bug.cgi?id=796745
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gst_curl_http_src_remove_queue_item() can free qelement and then
we get an invalid memory reference when we do qelement->next a
couple of lines below. Take the next pointer earlier so that we can
safely free.
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This should fix missing gstcameracalibrate.h and gstcameraundistort.h
missing into generated tarballs.
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This fixes an issue with SSA/ASS subtitles, where subtitles
would fail to appear if there was already a subtitle on screen.
This was because `struct _GstAssRender` had a single
`GstBuffer *subtitle_pending` member. This meant that
the assrender context could only be aware of one subtitle
at a time.
This patch changes the subtitle_pending member to a
linked list of pending subtitles.
The `gst_ass_render_chain_text` function no longer needs
to care about whether there are already subtitles pending,
it simply appends new subtitles to the list.
The `gst_ass_render_chain_video` function has been modified
to handle the list of pending subtitles.
Finally, the `gst_ass_render_pop_text` function has been
modified to pop the entire list of pending subtitles.
https://bugzilla.gnome.org/show_bug.cgi?id=735944
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When compiling with clang-6 this error raises:
raw_decoder.c:411:1: error: unused function 'cpr1204_crc'
[-Werror,-Wunused-function]
This patch only comments it out.
https://bugzilla.gnome.org/show_bug.cgi?id=796957
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When compiling with clang-6 this error pops out:
raw_decoder.c:1011:62: error: implicit conversion from enumeration
type 'const vbi_modulation' to different enumeration type
'vbi3_modulation' [-Werror,-Wenum-conversion]
This is because function vbi3_bit_slicer_set_params() sets
vbi3_modulation as enum type parameter, nonetheless vbi_modulation
enum is passed. Both enums looks semantically equal, thus the fix is a
simple cast.
https://bugzilla.gnome.org/show_bug.cgi?id=796957
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Needed for C++ headers. Without this, webrtcdsp can't find <algorithm>
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This is the native format that is in use by the webrtc audio processing
library internally, so this avoids internal {de,}interleaving and
format conversion (S16->F32 and back)
https://bugzilla.gnome.org/show_bug.cgi?id=793605
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Some SVG files created by inkscape use that, such as:
https://gitlab.gnome.org/GNOME/pitivi/uploads/8dd8d9d988b5eb6cc38f871196caac6f/Titel-Tafel3.2_anim.svg
This is a first patch for https://bugzilla.gnome.org/show_bug.cgi?id=796909
but it is not enough to support animated gifs.
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This uses the new path for OpenCV headers. OpenCV now have
master headers files per modules, which reduce the amount of
required includes. Note that HIGHGUI was included to get the
imgcodecs includes, which I fixed, though the master header is
missing the C headers, so I included that directly. All the
image stuff should be ported to C++ eventually. Finally, this
patch also update the header checks to reflect the modules that
are really being used.
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And removes all the ifdef code to support the 2.X APIs.
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This is to support older release of OpenCV2
https://bugzilla.gnome.org/show_bug.cgi?id=789075
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https://bugzilla.gnome.org/show_bug.cgi?id=789075
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Add support "srt://localhost:port" style uri, and change the
default host to "localhost"
https://bugzilla.gnome.org/show_bug.cgi?id=796842
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SRTClientSink is sender
https://bugzilla.gnome.org/show_bug.cgi?id=796842
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... and set SRT_EPOLL_IN flag if the client is not sender
(i.e., source element) since waiting readable event for
srt_recvmsg() makes more sense.
https://bugzilla.gnome.org/show_bug.cgi?id=796842
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... instead of doing it ourselves. Otherwise, we should add more
logic here (such as checking GstClock and etc) which was already provided by
GstBaseSrc.
https://bugzilla.gnome.org/show_bug.cgi?id=796842
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Given "passphrase" shouldn't be modified
https://bugzilla.gnome.org/show_bug.cgi?id=796842
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