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>
* RED_OR_ALPHA8 will map value to alpha for OpenGL, use R8 to avoid
2nd shader
* Determine texel size for proper texture memory preparation
* QByteArray::fromRawData() does shallow copy and thus leads to use of
corrupted memory
* Make sure RGBA dummy texture is fully opaque
* QRhiTexture::create() must be called to allocate texture resources
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6578>
If input height and parsed one are identical, do not consider it as interlaced
Fixing below pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420,width=640,height=10 \
! jpegenc ! jpegparse ! jpegdec ! videoconvert ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6181>
Apparently external-oes is not supported by the plugin as texture target,
while DMABuf uploading prefers it because it's zero copy.
This patch enables DMABuf uploading and rendering by using either 2D or
rectangle texture targets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5795>
The tags and caps were leaked for unknown streams, I'm not sure they'd be valid
in that case, but better safe than sorry.
The tags ownership is transfered when calling `gst_adaptive_demux_track_new()`
so unreffing those afterwards was a mistake.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5714>
In snapshot mode pngenc should output exactly one frame
and then return FLOW_EOS to upstream. If upstream sends
more input frames before shutting down, it should keep
returning FLOW_EOS but not output any more encoded frames.
After a flushing seek it should output frames again though.
Fixes#3069.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5546>
After rendering a QML scene the qml6glsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.
This commit waits for the sync point inside the qml6glsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.
This is a port of the original fix for the qmlglsrc element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5519>
After rendering a QML scene the qmlglsrc element copies the contents of
the scene to a GStreamer buffer. This happens on the Qt render thread.
Then it attaches a sync point to the destination buffer. This sync point
must be awaited by other threads which use the buffer later on. The
current implementation relies on the downstream elements to wait for the
sync point. However, there are situation where this does not work. The
GstBaseTransform e.g. copies the buffer metadata (which overwrites the
sync point without waiting for it) *before* waiting for the sync point.
This commit waits for the sync point inside the qmlglsrc element before
sending it downstream. The wait command is issued on the streaming
thread with the pipeline OpenGL context, i.e. it will synchronize with
the GStreamer OpenGL thread.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5506>
There is a race condition where transfer has not been submitted yet while the
request is cancelled which leads to the transfer state going back to
`DOWNLOAD_REQUEST_STATE_OPEN` and the user of the request to get signalled about
its completion (and the task actually happening after it was cancelled) leading
to assertions and misbehaviours.
To ensure that this race can't happen, we start differentiating between the
UNSENT and CANCELLED states as in the normal case, when entering `submit_request`
the state is UNSENT and at that point we need to know that it is not because
the request has been cancelled.
In practice this case lead to an assertion in
`gst_adaptive_demux2_stream_begin_download_uri` because in a previous call to
`gst_adaptive_demux2_stream_stop_default` we cancelled the previous request and
setup a new one while it had not been submitted yet and then got a `on_download_complete`
callback called from that previous cancelled request and then we tried to do
`download_request_set_uri` on a request that was still `in_use`, leading to
something like:
```
#0: 0x0000000186655ec8 g_assert (request->in_use == FALSE)assert.c:0
#1: 0x00000001127236b8 libgstadaptivedemux2.dylib`download_request_set_uri(request=0x000060000017cc00, uri="https://XXX/chunk-stream1-00002.webm", range_start=0, range_end=-1) at downloadrequest.c:361
#2: 0x000000011271cee8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_begin_download_uri(stream=0x00000001330f1800, uri="https://XXX/chunk-stream1-00002.webm", start=0, end=-1) at gstadaptivedemux-stream.c:1447
#3: 0x0000000112719898 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment [inlined] gst_adaptive_demux2_stream_download_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:0
#4: 0x00000001127197f8 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_load_a_fragment(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:1969
#5: 0x000000011271c2a4 libgstadaptivedemux2.dylib`gst_adaptive_demux2_stream_next_download(stream=0x00000001330f1800) at gstadaptivedemux-stream.c:2112
```
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5435>
The counter was using a signed 8 bit integer, which was overflowing
after 127 entries. That was then passed as an unsigned 32 bit integer to
libflac, which caused it to be converted to a huge unsigned number.
That then caused an invalid memory access inside libflac.
As a bonus, signed integer overflow is undefined behaviour.
Instead, use an unsigned 8 bit integer. Once this overflows the existing
code already catches it and stops adding the cue. While FLAC__metadata_object_cuesheet_insert_track()
takes an unsigned 32 bit integer for the track number, FLAC__StreamMetadata_CueSheet_Track is
limiting it to an unsigned 8 bit integer.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2921
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5420>
Issue was that Qt was caching multiple different shadersbased on a static
variable location. This static variable location was the same no matter
the input video format and so if an item had an input video format of
RGB and another of RGBA, they would eventually end up using the same
GL shader leading to incorrect colours.
Fix by providing different static variable locations for each of the
different shaders that will be produced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5145>
This check fixes a critical warning that can happen when a pointer motion
happens and the video doesn't have its width/height information available.
GStreamer-Video-CRITICAL **: gst_video_center_rect: assertion 'src->h != 0' failed
#0 g_logv (log_domain=0x7ffff705e176 "GStreamer-Video", log_level=G_LOG_LEVEL_CRITICAL, format=<optimized out>, args=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1422
#1 0x00007ffff7e1a81d in g_log (log_domain=<optimized out>, log_level=log_level@entry=G_LOG_LEVEL_CRITICAL, format=format@entry=0x7ffff7e77a9d "%s: assertion '%s' failed") at ../../../../Projects/jhbuild/glib/glib/gmessages.c:1460
#2 0x00007ffff7e1b749 in g_return_if_fail_warning (log_domain=<optimized out>, pretty_function=<optimized out>, expression=<optimized out>) at ../../../../Projects/jhbuild/glib/glib/gmessages.c:2930
#3 0x00007ffff701d90b in gst_video_sink_center_rect (src=..., dst=..., result=result@entry=0x7fffffffc6d0, scaling=scaling@entry=1) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-base/gst-libs/gst/video/gstvideosink.c:105
#4 0x00007fffe5652dbb in _fit_stream_to_allocated_size (result=0x7fffffffc6d0, allocation=0x7fffffffc6c0, base_widget=0x9396f0) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:326
#5 gtk_gst_base_widget_display_size_to_stream_size (base_widget=base_widget@entry=0x9396f0, x=1207.7109375, y=811.84765625, stream_x=stream_x@entry=0x7fffffffc720, stream_y=stream_y@entry=0x7fffffffc728) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:344
#6 0x00007fffe5651a4b in gst_gtk_base_sink_navigation_send_event (navigation=0x5ff990, event=0x178a730) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gstgtkbasesink.c:340
#7 0x00007fffe5652432 in gtk_gst_base_widget_motion_event (widget=<optimized out>, event=event@entry=0x1f14b60) at ../../../../Projects/jhbuild/gstreamer/subprojects/gst-plugins-good/ext/gtk/gtkgstbasewidget.c:404
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5049>
The libpsl subproject wasn't building successfully and CI didn't
notice because:
1. The plugin wasn't explicitly enabled
2. Even when the plugin is explicitly enabled, the dep is not required
at build time when not building a static plugin
So fix all of these issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5038>