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>
By this commit, following formats will be newly supported by d3d11 elements
* Y444_{8, 12, 16}LE formats:
Similar to other planar formats. Such Y444 variants are not supported
by Direct3D11 natively, but we can simply map each plane by
using R8 and/or R16 texture.
* P012_LE:
It is not different from P016_LE, but defining P012 and P016 separately
for more explicit signalling. Note that DXVA uses P016 texture
for 12bits encoded bitstreams.
* GRAY:
This format is required for some codecs (e.g., AV1) if monochrome
is supported
* 4:2:0 planar 12bits (I420_12LE) and 4:2:2 planar 8, 10, 12bits
formats (Y42B, I422_10LE, and I422_12LE)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2346>
Some elements will make use of the automatically generated names to
create new pads in future muxer instances, for example splitmuxsink.
Previously we would've created a pad with a random pid that would become
"sink_0", and then on a new muxer instance a pad "sink_0" and tsmux
would've then failed because 0 is not a valid PID.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2318>
When B-frame is enabled, encoder seems to adjust PTS of encoded sample
by using frame duration.
For instance, one observed timestamp pattern by using B-frame enabled
and 30fps stream is:
* Frame-1: MF pts 0:00.033333300 MF dts 0:00.000000000
* Frame-2: MF pts 0:00.133333300 MF dts 0:00.033333300
* Frame-3: MF pts 0:00.066666600 MF dts 0:00.066666600
* Frame-4: MF pts 0:00.099999900 MF dts 0:00.100000000
We can notice that the amount of PTS shift is frame duration and
Frame-4 exhibits PTS < DTS.
To compensate shifted timestamp, we should
calculate the timestamp offset and re-calculate DTS correspondingly.
Otherwise, total timeline of output stream will be shifted, and that
can cause time sync issue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2354>
Previously pads might have been requested already (e.g. in NULL state),
then reset was called (e.g. because changing state) and then a new pad
was requested. Resetting is re-creating the internal muxer object and as
such resetting the pid counter, so the next requested pad would get the
same pid as the first requested pad which then leads to collisions.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2317>
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>
As per SDK doc, IDeckLinkInputCallback::VideoInputFrameArrived
method might not provide video frame and it can be null.
In that case, given stream_time can be invalid.
So, we should not try to convert GST_CLOCK_TIME_NONE
by using gst_clock_adjust_with_calibration()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2337>
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>
Track kernel and VA driver dependencies so gstreamer
will re-inspect the plugin if any of them change.
Also, do not blacklist the plugin if !msdk_is_available
since it could be a transient issue caused by one or
more external dependency issues (e.g. wrong/missing
driver specified, but corrected by user later on).
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2335>
Some decoder implementations might drain out internal buffers and
reset its status on segment-done event. So, in case that
upstream stream-format is packetized but downstream supports only
byte-format, required codec-data might not be forwarded toward
downstream if such parameter set NAL units don't exist in inband
bitstream. Therefore, parse elements should re-send parameter set NAL
units like the case of flush event.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2334>
We add 12 bits entries into this default mapping. And the old mapping
is not precise. For example, the NV12 should not be used as the default
mapping for VA_RT_FORMAT_YUV422 and VA_RT_FORMAT_YUV444, it is even not
a 422 or 444 format.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2332>
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>
For those using context from the application which
would be the embedded video case, if the frame callback
is entering at the same time as window is finalizing,
a wayland proxy object would be destroyed twice, leading
the refcout less than zero in the second time, it can
throw an abort() in wayland.
For those top window case, which as a directly connection
to the compositor, they can stop the message queue then
the frame callback won't happen at the same time as the
window is finalizing. It doesn't think it would bother
them about this.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1883>
alphacombine element is a simple element that assumes buffers are always
paired, or at least that missing buffers are signalled with a GAP. The QoS
implementation in the GstVideoDecoder base class allow decoders dropping
frames independently and that could lead to stall in alphacombine.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2326>
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>
