In qml6glsrc, we capture the application by copying the back buffer into
our own FBO. The afterRendering() signal is too soon as from the apitrace, the
application has been rendered into a QT internal buffer, to be used as a cache
for refresh.
Use afterFrameEnd() signal instead. This works with no delay on GLES. With GL
it seems to reduce from 2 to 1 frame delay (this may be platform specific). A
different recording technique would need to be used to completely remove this
delay.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7351>
By setting the earliest time to timestamp + 2 * diff there would be a difference
of 1 * diff between the current clock time and the earliest time the element
would let through in the future. If e.g. a frame is arriving 30s late at the
sink, then not just all frames up to that point would be dropped but also 30s of
frames after the current clock time.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7459>
When no ports are given, gst_jack_get_ports() is called to get all the
(physical) output ports but then the result is ignored, triggering the
"No physical output ports found..." error.
Instead, move the queried ports to the variable we're going to use
later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7474>
```
In file included from ../subprojects/gst-plugins-good/ext/qt6/gstqsg6material.cc:31:
../subprojects/gst-plugins-good/ext/qt6/gstqsg6material.h:69:17: error: private
field 'mem_' is not used [-Werror,-Wunused-private-field]
69 | GstMemory * mem_;
| ^
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7414>
With GLES 2.0 we are forced to use CopyTextImage2D which requires
passing an internal format. With QT6 eglfs, we need to pass GL_RGB
instead, probably because of how the texture has been created. As its
hard to guess, simply fallback to GL_RGB on failure. This fixes usage
or qml6glsrc with eglfs backend, without loosing support for
semi-transparent window on other platforms.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7321>
In order to use oes-external, the qml6glsink needs a fragment shader that uses
the samplerExternalOES.
The qsb tool is not able to handle shaders that contain samplerExternalOES since
this feature is not supported by all target shading languages. The qsb tool is
able to replace a shader in the qsb file to handle this use case. Use it to
generate a shader variant that uses samplerExternalOES for OpenGL ES and select
that variant if the qml6glsink negotiated texture target oes-external.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7319>
The initial goal was to support the case where we are paused watching a live
stream, and when we resume we can no longer resume from the previously
downloaded position. In that case we internally do a flushing seek back to the
"current live head position". This was also extended since to be able to
handle (utterly broken) servers when we can't really figure out where we are
anymore and therefore trigger that lost sync so we can try to get back on our
feet.
This does fix the issue... but results in spurious FLUSH_{START|STOP} events
being sent downstream. While that's fine for regular playback scenarios, it's a
bit of a wild scenario since a lot of pipelines/applications don't expect such
events when it wasn't triggered by downstream/application.
Fixes#3605
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7005>
This attempts to implement the gtkglsink element on Windows using WGL,
as there were some more gotchas that are along the way, since we need to
juggle with libepoxy along the way, meaning that we need a recent
GTK+-3.24.x for this to work properly, i.e. the upcoming GTK+-3.24.43.
Since we are essentially using an overlay compositor only during
rendering, move its initialization and destruction into the
gtk_gst_gl_widget_render() function, so that things are safer as we are
doing things across threads between gstreamer (gst-gl) and GTK, as GL
operations, as above, have more gotchas on Windows.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4289>
Some servers might not provide 100% matching PDT when doing updates, or accross
variants. This would cause the code matching segments using PDT to fail if the
segment PDT was 1 microsecond (or whatever small value) before the candidate
segment. And would pick the (wrong) following segment as the matching one.
In order to be more tolerant when matching, we instead check whether the
candidate segment is within the first segment of the segment we are trying to
match.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
If we end up with a segment with an internal time that varies from the supposed
one, this could be for two reasons:
* We guess-timated the wrong segment to go to when advancing or switching
variants. In that case we try to find the actual segment to go to (just before
this change).
* There was a complete playlist change (for whatever reason) and we can't find a
replacement. In that case we want to carry on playback from this position but
need to remember that we moved (by setting the stream to DISCONT, and
resetting the new mapping).
Fixes playback on several broken stream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
Since the default value of `m3u8->discont_sequence` (before parsing of the
playlist data) was 0 .. we would never properly detect the presence of that
field if it was present with a value of 0.
This would later on cause havoc in playlist synchronization where we would
assume it didn't have a discontinuity sequence specified (whereas it did, and it
was 0).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
A lot of streams will do a poor job of estimating proper duration of fragments
in the playlist, but over several fragments have it correct.
Instead of constantly trying to realign the estimated stream time, allow for a
more realistic tolerance of 3-4 video frames
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
When updating playlists, we want to know whether the updated playlist is
continuous with the previous one. That is : if we advance, will the next
fragment need to have the DISCONT buffer set on it or not.
If that happens (because we switched variants, or the playlist all of a sudden
changed) we remember that there is a pending discont for the next fragment. That
will be used and resetted the next time we get the fragment information.
Previously this was only partially done. And it was racy because it was set
directly on `GstAdaptiveDemux2Stream->discont` when a playlist was updated,
instead of when the next fragment was prepared.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
When dealing with live streams, the function was assuming that all segments of
the playlist had valid stream_time. But that isn't TRUE, for example in the case
of failing to synchronize playlists.
Fixes losing sync due to not being able to match playlist on updates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
This counter is incremented once for every segment, meaning it would
e.g. overflow after 24 days when using 1ms segments. Once that happens,
completely wrong positions are reported and invalid memory is handed out
for writing/reading the next segments.
As the affected variables are unfortunately part of the public API of
the struct, a second set of variables is added together with accessor
functions and both variables are kept in sync for backwards
compatibility.
All existing users of the two variables are moved to the new ones but
external code might still run into the overflow.
This also slightly breaks API as external code updating the variables
will have no effect anymore but the only known user of this is
pulsesink.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6740>
Set as much information as possible on the slot (including the associated
track) *before* the associated source pad is added to the element.
We need this so that incoming event/queries can be replied to if they are
received when adding the pad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6690>
