GstRTPHeaderExtension::write can map the RTP buffer for reading. If that
happens on a buffer that is already mapped WRITE-only by the payloader,
the payloader's mapping gets invalidated (GstRTPBuffer::map will point
to a different instance of GstMemory).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1173>
Usually the latency message is only posted whenever latency of an
element changes but that might be too early as the sinks might not be
able to query the latency at that point yet.
Similarly adding a new sink should cause latency reconfiguration once
that new sink is able to report its latency.
This fixes latency configuration in pipelines where webrtcbin is the
only "sink", i.e. it is used in a sendonly session. Before, the latency
would always be configured to 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/843>
The finish() virtual function documentation state that "Sub-classes can refuse
to decode new data after." Though, it is very common to issue a non-flushing
seek after that event in gapless playback uses case. This fixes potential
stalls with code using segment seeks, by using drain() virtual funciton
instead.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1206>
There may be two or more threads involved here however the important
interaction is the use of ogg->seeK_event_drop_till value that was only
set in the push-mode seek-event thread and could race with upstream
sending e.g. and EOS (or data).
Scenario is this:
1. oggdemux performs a seek to near the end of the file to try and find
the duration. ogg->push_state is set to PUSH_DURATION.
2. Seek is picked up by the dedicated seek event thread and sets
ogg->seek_event_drop_till to the seek event's seqnum.
3. Most operations are blocked or dropped waiting on the duration to
be determined and processing continues until a duration is found.
4. Two branching options for how this ultimately plays out
4a. The source is too fast and we receive an EOS event which is dropped
because ogg->push_state == PUSH_DURATION. In this case everything
works.
4b. We hit our 'almost at the end' check in
gst_ogg_pad_handle_push_mode_state() and attempt to seek back to the
beginning (or to a user-provided seek). This seek is marshalled to
the seek event thread without setting ogg->seek_event_drop_till but
with change ogg->push_state = PUSH_PLAYING. If an EOS event or
e.g. buffers arrive from upstream before the seek event thread has
picked up the seek event, then the EOS/data is processed as if it
came as a result of the seek event. This is the case that fails.
The fix is two-fold:
1. Preemptively set ogg->seek_event_drop_till when setting the seek
event so that data and other events can be dropped correctly.
2. In addition to dropping and EOS events while ogg->push_state ==
PUSH_DURATION, also drop any EOS events that are received before the
seek event has been processed by also tracking the seqnum of the seek.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1196>
If downstream tries to seek in BYTES format, don't pass that through
to upstream. The byte positions downstream requests won't make any
sense in the muxed stream. There might be other formats we want to
pass through to upstream, but BYTES is not one of them. If we get a
seeking query about BYTES format, refuse that too.
This fixes a situation where we're playing a fragmented mp4 over http
and qtdemux refuses the initial seek (in TIME format), but then
h264parse/baseparse send a seek in BYTES format and everything falls
apart.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1014>
If a property is supplied to gst-launch-1.0 to set on a property that
implements GstChildProxy, it would always accept any property name
and try to set it later. This means that (for example) decodebin
will accept and not complain about property names that can never exist like:
gst-launch-1.0 videotestsrc ! decodebin NON-EXISTING_PROPERTY=adsfdasf ! fakesink
Instead, only try to do deferred property setting for property names
that contain the :: separator that indicates it's a setting on a child
that might appear later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/832>
Updating of codec_data in the caps is important to propagate changes
in sps/pps/vps via NALs. Without this, downstream does not renegotiate
when upstream changes resolution.
The comment referring to rtph264pay is from 2015 and is out of date.
rtph264pay stopped doing that in 2017 with commit
dabeed52a9
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/1011>
below commit change the window resize thread and cause viv-fb backend
hang, need move resize code after window->open is called. Otherwise,
the resize message will send to a thread that not start running and
window resize call will waiting forever.
Commit: b887db1efe
glwindow: fix racy resize updates
Take locks around resize handling and marshall all resizes to the
windowing thread by default.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1195>
If GST_GL_WINDOW is unset but GST_GL_PLATFORM=egl, then we were choosing
to create an GstGLDisplayEGL directly instead of going through the any
more specific windowing system implementation (X11, Wayland).
The 'create an GstGLDisplayEGL when GST_GL_PLATFORM=egl' was a fallback
as we did not have entries for all EGL-using window systems previously.
Now that we do, the fallback can be removed. An EGLDisplay can still
be created by setting GST_GL_WINDOW=egl or as one option.
Fixup of https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1154
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1169>
- atom nodes/bytereader sizes are already checked
- palettes: are fixed/known size
g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.
Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>
- ebml-read: add some sanity checks when going from 64-bit
to 32-bit length
- matroska-ids: codec_data_size has been checked via
gst_ebml_read_binary(), is existing allocation.
- matroska-demux: alloc size is from existing allocations
g_memdup() is deprecated since GLib 2.68 and we want to avoid
deprecation warnings with recent versions of GLib.
Also use gst_buffer_new_memdup() instead of _wrapped(g_memdup(),..).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/993>
