Implement 8-bit values of SMPTE RP 2019-1:2014. The bar widths and
heights are the result of fractions as integers. The remainders of
widths are distributed in a way that they match the values in Table
C.1 (a) in the specification.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/-/merge_requests/1063>
We are dropping RASL (Random Access Skipped Leading picture) which
is associated with an IRAP (Intra Random Access Picture) that has
NoRaslOutputFlag equal to 1, since the RASL picture will not be
outputted and also it should not be used for reference picture.
So, corresponding GstVideoCodecFrame should be released immediately.
Otherwise GstVideoDecoder baseclass will hold the unused frame.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2330>
We call gst_vaapiencode_drain() in gst_vaapiencode_change_state(),
whose context does not hold the stream lock of the encoder. The
current gst_vaapiencode_drain inside unlock/lock pair adds a extra
lock count to the stream lock of encoder and causes hang later.
We just remove the gst_vaapiencode_drain() and expand its logic
correctly according to the lock/unlock context.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer-vaapi/-/merge_requests/433>
The build fails on macos with the following error:
/usr/local/Cellar/opencv/4.5.0_5/include/opencv4/opencv2/core/mat.hpp:2226:15: error: no template named 'initializer_list' in namespace 'std'
Mat_(std::initializer_list<_Tp> values);
fatal error: too many errors emitted, stopping now [-ferror-limit=]
35 warnings and 20 errors generated.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2368>
If they're not dropped, they can be blocked in the queue even if it is
leaky in the case where there is a buffer being pushed downstream. Since
in webrtc, it's unlikely that there will be a special allocator to
receive RTP packets, there is almost no downside to just ignoring the
queries.
Also drop queries if they get caught in the pad probe after the queue.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2363>
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>
By using the clocksync inside the dtlssrtpenc, all streams inside a
bundled are synchronized together. This will cause problems if their
buffers are not already arriving synchronized: clocksync would wait for
a buffer on one stream and then buffers from the other stream(s) with
lower timestamps would all be sent out too late.
Placing the clocksync before the rtpbin and rtpfunnel synchronizes each
stream individually and they will be send out more smoothly as a result.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2355>
If the decoder has crop_top/left value > 0(e.g. the conformance
window in the H265). Which means that the real output picture
locates in the middle of the decoded buffer. If the downstream can
support VideoCropMeta, a VideoCropMeta is added to notify the
real picture's coordinate and size. But if not, we need to copy
it manually and the other_pool is needed. We always assume that
decoded picture starts from top-left corner, and so there is no
need to do this if crop_bottom/right value > 0.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
The base va decoder's video_align is just used for calculation the
real decoded buffer's width and height. It does not have meaning
for the VideoMeta, because it does not align to the real picture
in the output buffer. We will use VideoCropMeta to replace it later.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
The base va decoder's video_align is just used for calculation the
real decoded buffer's width and height. While the gst_video_info_align
just calculate the offset and stride based on the video_align. But
all the offsets and strides are overwritten in gst_va_dmabuf_allocator_try
or gst_va_allocator_try, which make that calculation useless.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2298>
