Receiving the WEBKIT_LOAD_COMMITTED event doesn't actually
mean we have committed an SHM buffer / image yet.
As this is the condition we are interested in, check it
instead.
Also wrap g_cond_wait in a loop for extra correctness points.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1476>
"waylandsink: use GstMemory instead of GstBuffer for cache lookup"
changes the cache key to GstMemory, but the cached data still needs
a pointer to the GstBuffer to control the buffer lifecycle.
If the GstMemory used as the cache key moves from one GstBuffer to
another, the pointer in the cached data will be out-of-date.
Update the current GstBuffer pointer for each frame so that it always
represents the currently in use (from attach to release) GstBuffer
for each wl_buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1473>
The GstMemory objects contained in a GstBuffer could be replaced
by an upstream element, which would break the association beteen
the GstBuffer and the wayland wl_buffer, make the cache lookup
results incorrect.
This patch changes the cache lookup to use the first GstMemory
in a buffer instead. For multi-plane buffers, this assumes that
all of the GstMemory(s) will always be moved together as a set,
and that the same (first) GstMemory isn't used with different
combinations of other GstMemory(s).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1401>
Instead of attaching a single wayland wl_buffer to each GStBuffer as qdata,
keep a separate cache for each display.
A unique wl_buffer and associated metadata is created for each display.
This allows for sharing of GstBuffer objects between multiple
displays, such as when using tee elements.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1401>
A pipeline like this:
closedcaption/x-cea-708,format=cdp,framerate=30000/1001 ! ccconverter ! closedcaption/x-cea-708,format=cc_data
would produce a critical/assert:
GStreamer-CRITICAL **: 14:21:11.509: gst_util_fraction_multiply: assertion 'a_d != 0' failed
because there would be no framerate field on ccconverter's output.
Fixed by always fixating a framerate if the input has a framerate.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1393>
Unclear why hotdoc wants 'gstavtp' as the plugin name here,
that's just wrong.
Add since marker and mark private subclasses as plugin API
so hotdoc knows they belong to the plugin and aren't external.
Fix GstAvtpAafTstampMode get_type() function.
keys_exported flag should be set only if keys are actually exported.
For that the next conditions are needed:
1 - SSL_export_keying_material on success
2 - SSL_get_selected_srtp_profile returns a valid profile
3 - The profile ID is SRTP_AES128_CM_SHA1_80 or SRTP_AES128_CM_SHA1_32
Also don't crash if NULL is returned as profile.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1156>
SURROUND is more to spec according to the FIXME comments, so add this.
Also add SIDE for 5 and 5.1 because of ffmpeg compatibility, because the
following pipeline downmixes to mono otherwise:
gst-launch-1.0 audiotestsrc num-buffers=1 ! audio/x-raw, channels=6 !
avenc_ac3 ! avdec_ac3 ! audioconvert ! fdkaacenc ! fakesink -v
Fixes#1327
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1352>
Unfortunately it means those tune enums don't show up in
the docs then, but if that's how it's gotta be..
(Problem at hand is that on Tim's machine x265enc gets an
tune=animation and on the CI machine this doesn't show up.)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1354>
The unit tests only checked for vulkan_dep.found(), which can
be true if the libs are there but glslc was not found, in which
case the plugin wouldn't be built and the unit tests would fail
because of missing vulkan plugins.
Doesn't really make much sense to build the vulkan integration lib
either if we're not going to build the vulkan plugin, so just disable
both for now if glslc is not available.
Fixes#1301
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1307>
This is needed for cross-compiling without a build machine compiler
available. The option was added in 0.54, but we only need this in
Cerbero and it doesn't affect older versions so it should be ok.
Will only cause a spurious warning.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1266>
Instead of storing the raw cc_data, store the 2 cea608 fields individually
as well as the ccp data.
Simply copying the input cc_data to the output cc_data violates a number of
requirements in the cea708 specification. The most prominent being, that
cea608 triples must be placed at the beginning of each cdp.
We also need to comply with the framerate-dpendent limits for both the
cea608 and the ccp data which may involve splitting or merging some
cea608 data but not ccp data or vice versa.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1116>
Add some properties to allow TCP and UDP candidates to be toggled. This
is useful in cases where someone is using this element in an environment
where it is known in advance whether a given transport will work or not
and will prevent wasting time generating and checking candidate pairs
that will not succeed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1223>
When negotiating the SDP we should only connect the streams that are
actually mentioned in the SDP. All other streams are not relevant at
this time and would likely be part of a future SDP update. Fixes a
couple of the renegotiation webrtc unit tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1240>
Proper calculate running time for buffers that are out of current
segment and try to honor them.
A typical case is for AVTP packets coming from avtpcvfpay element, as
those may have DTS that falls out of segment (which is about PTS).
By using gst_segment_to_running_time_full(), avtpsink can properly
calculate when to transmit those buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
Seek events will cause new segments to be sent to avtpcvfpay, and for
flushing seeks, a pipeline running time reset. This running time
reset, which effectively changes pipeline base time, will cause
avtpcvfpay element to generate incorrect DTS for the initial set of
buffers sent after FLUSH_STOP.
This happens due the fact that base time change happens only when the
sink gets the first buffer after the FLUSH_STOP - so avtpcvfpay used
the wrong base time to do its calculations.
However, if the pipeline is paused before the seek, sink will update
base time when pipeline state goes to PLAYING again, before avtpcvfpay
gets the first buffers after the flush. Then avtpcvfpay element will be
able to normally calculate DTS for the outgoing packets.
This patch simply adds a warning message in case a flushing seek is
performed on a playing pipeline.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
TSN streams are expected to send packets to the network in a well
defined "pace", which is arbitrarily defined for each stream. This pace
is defined by the "measurement interval" property of a stream.
When the AVTP CVF payloader element - avtpcvfpay - fragments a video
frame that is too big to be sent to the network, it currently defines
that all fragments should be transmitted at the same time (via DTS
property of GstBuffers generated, as sink will use those to time the
transmission of the AVTPDU). This doesn't comply with stream definition,
which also has a limit on how many packets can be sent on a given
measurement interval.
This patch solves that by spreading in time the DTS of the GstBuffers
containing the AVTPDUs. Two new properties, "measurement-interval" and
"max-interval-frames", added to avptcvfpay element so that it knows
stream measurement interval and how many AVTPDUs it can send on any of
them. More details on the method used to proper spread DTS/PTS according
to measurement interval can be found in a code commentary inside this patch.
Tests also added for the new property and behaviour.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1004>
If the remote is bundling, but we are not and remote is offering.
we cannot put the remote media sections into a bundled transport as that
is not how we are going to respond.
This specific failure case was that the remote ICE credentials were
never set on the ice stream and so ice connectivity would fail.
Technically, this whole bunde-policy=none handling should be removed
eventually when we implement bundle-policy=balanced. Until such time,
we have this workaround.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1231>
