Limitations:
- No transport changes at all (ICE, DTLS)
- Codec changes are untested and probably don't work
- Stream removal doesn't remove transports (i.e. non-bundled transports
will stay around until webrtcbin is shutdown)
- Unified Plan SDP only. No Plan-B support.
We add the signal watch in testSeekPreTestCallback so
remove it in testSeekPostTestCallback and not deep inside
some if clause in some other callback somewhere.
Based upon the souphttpsrc tests, add unit tests for the curlhttpsrc
element. The souphttpsrc tests are able to use an HTTP server that
is provided as part of the soup library. This does not exist in the
curl library, therefore these tests provide a very simple HTTP server
using the GIO library.
These curlhttpsrc tests contain one new test that does not come from
the souphttpsrc tests. The test_multiple_http_requests test tries to
reproduce the way in which GstAdaptiveDemux makes use of URI source
elements. GstAdaptiveDemux creates a bin with the httpsrc element
and a queue element and sets the locked state of that bin to TRUE,
so that it does not follow the state transitions of its parent. It
then moves this bin to the PLAYING state to start each download and
back to READY when the download completes.
The VCD source was ported in 2014 (commit 89eb1e9), but the necessary
"cdxaparse" plugin, which is used to "Parse a .dat file (VCD) into
raw mpeg1" was never ported.
This means that the probable main user for the feature, totem, hasn't
actually been able to play back VCDs, since 2012, when it switched to
using GStreamer 1.0.
Note that even if cdxaparse was finally ported, a lot of work would
still be necessary before it is considered usable. Notably, it is
missing disc image support [1] and some VCDs just cannot be opened for
reading [2].
[1]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/898
[2]: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/899
Allow fallback to orc subproject if any, and add missing orc version check.
Additionally 'dependencies' keyword is removed from find_library,
because it's invalid keyword for find_library.
Allow run some unit tests on Windows.
* Add dependency explicitly for some test cases, otherwise plugins couldn't be
loaded on uninstalled environment of Windows.
* Add missing GST_PLUGIN_LOADING_WHITELIST on meson build.
It depends on the framerate how many cc_data byte pairs are allowed per
frame, and the framerate is also needed for converting into the CDP or
MCC format as the framerate is part of the header metadata.
This is for the same reason as the dash tests. This should ideally
be converted to gst-validate tests. These tests randomly timeout also
due to the tests doing seeks from the streaming thread (sic).
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 works like a valve in front of the actual avwait. When recording ==
TRUE, other rules are then examined. When recording == FALSE, nothing is
passing through.
https://bugzilla.gnome.org/show_bug.cgi?id=796836
We used to have the same enum to represent H265 profiles and idc values.
Those are no longer the same with extension profiles defined from
version 2 of the spec.
Split those enums so the semantic of each is clearer and we'll be able
to add extension profiles to GstH265Profile.
Also add gst_h265_profile_tier_level_get_profile() to retrieve the
GstH265Profile from the GstH265ProfileTierLevel. It will be used to
implement the detection of extension profiles.
https://bugzilla.gnome.org/show_bug.cgi?id=793876
SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
Try prioritizing downstream's caps over upstream's if possible so the
parser can configured in "passthrough" if possible and save it from
doing useless conversions.
https://bugzilla.gnome.org/show_bug.cgi?id=790628
Unfortunately we need to use an extra set of parenthesis for each data level.
For details see:
https://gcc.gnu.org/bugzilla/show_bug.cgi?id=53119
Affected versions are e.g.
gcc (Ubuntu 4.8.4-2ubuntu1~14.04.3) 4.8.4
which is the default on ubuntu-trusty. I looks like the fix was never
backported.
Remove gst_init() from a few tests. Use _OBJECT variants in logging. Remove
arbitrary extra blank lines. Make push_event() more like push_buffer() - set
the event to NULL and add cleanup to _chain_data_clear().
Using two (or more) probes on the same pad where one of the probe
returns HANDLED or DROP is tricky since the other probes might
not be called.
Instead use regular probes and a proper pad (the sinkpad already existed,
it only required to be activated and have a dummy chain function for
the events/buffers to be received/handled properly)
In most cases we want to stop the pipeline just once, but we have
to do this from code that runs in the streaming threads and in case
we have multiple streams, we need to make sure that we do this only
once. The previous checks were broken, this should fix it.
https://bugzilla.gnome.org/show_bug.cgi?id=786006
Except for gst/gl/gstglfuncs.h
It is up to the client app to include these headers.
It is coherent with the fact that gstreamer-gl.pc does not
require any egl.pc/gles.pc. I.e. it is the responsability
of the app to search these headers within its build setup.
For example gstreamer-vaapi includes explicitly EGL/egl.h
and search for it in its configure.ac.
For example with this patch, if an app includes the headers
gst/gl/egl/gstglcontext_egl.h
gst/gl/egl/gstgldisplay_egl.h
gst/gl/egl/gstglmemoryegl.h
it will *no longer* automatically include EGL/egl.h and GLES2/gl2.h.
Which is good because the app might want to use the gstgl api only
without the need to bother about gl headers.
Also added a test: cd tests/check && make libs/gstglheaders.check
https://bugzilla.gnome.org/show_bug.cgi?id=784779
Sending an event can accepted event if the caps were rejected
because the event could be queued and processed later.
Also send a drain query in the caps test to make sure that the
event has been processed.
https://bugzilla.gnome.org/show_bug.cgi?id=781673
Since insertion of aud landed, we need to change some testcases
accroding to the change.
Note that counting frames are changed in parser.c,
due to generated frames, AUD.
https://bugzilla.gnome.org/show_bug.cgi?id=736213
For duration queries on live streams, adaptivedemux ignores the query.
The problem then is that the query is answered by the downstream
qtdemux element, with the duration of the currently passing fragment.
This commit changes the behaviour of adaptivedemux to answer the duration
queries for live streams, returning GST_CLOCK_TIME_NONE.
https://bugzilla.gnome.org/show_bug.cgi?id=753879
See https://bugzilla.gnome.org/show_bug.cgi?id=773666
This would ideally be solved in baseparse but that requires further
thought at this point, and in the meantime it would be good to have
rawbaseparse not assert on this but handle it gracefully instead.
Make the unit tests handle the fact that pads don't appear
immediately. Before, the test assumed pads are exposed before the
internal source element is created, which is no longer true.
To satisfy follwing restriction of HLS spec 6.3.3,
select startup fragment sequence to 4th from end of playlist.
Also, seek range should exclude last three fragment in playlist.
"the client SHOULD NOT choose a segment which starts less than
three target durations from the end of the Playlist file."
https://bugzilla.gnome.org/show_bug.cgi?id=777682
