https://bugzilla.gnome.org/show_bug.cgi?id=741398 changed
rtpptdemux in 2014 to not post a GST_ELEMENT_ERROR on the
bus when dropping an invalid (non-RTP) packet, but still
returned GST_FLOW_ERROR upstream - so the pipeline still
stops, but now without a useful bus error.
Return GST_FLOW_OK instead, so the pipeline keeps
running. Some old telephony equipment can send invalid
packets before the real RTP traffic starts.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2523>
get_colorspace() checks input caps transfer when mapping V4L2_XFER_FUNC_709
back to V4L2_COLORSPACE_BT2020 and GST_VIDEO_TRANSFER_BT2020_12. After
receiving source change event, decoder will G_FMT and S_FMT again. So need
to reset transfer when acquiring format to avoid using the old transfer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2505>
In case of per features registration such as the
customizable gstreamer-full library, each
element should check that the soup library can be loaded to
facilitate the element registration.
Initialize the debug category properly
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2491>
zlib is required, and if it isn't found it is checked several ways and
then forced via subproject(). This code was added in commit
b93e37592a, to account for systems where
zlib doesn't have pkg-config files installed.
But Meson already does dependency fallback, and also, since 0.54.0, does
the in-between checks for find_library('z') and has_header('zlib.h') via
the "system" type dependency. Simplify dependency lookup by marking it
as required, which also makes sure that the console log doesn't
confusingly list "not found".
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2486>
- Consistently unref the chained buffer at the end of the chain
function, if we're not handing it off to `gst_pad_push`. This avoids a
few buffer leaks in the error paths in `_chain` and `_push_history`.
- When mapping the video frame fails, return a flow error instead of
crashing.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2444>
If for some reason the encoder produces frames with a pts higher than
the input one, we were dropping all the video encoder frames and ended
up crashing when trying to access the pts of a NULL pointer returned by
gst_video_encoder_get_oldest_frame().
I hit this scenario by feeding a decreasing timestamp to vp8enc which
seem to confuse the encoder.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2436>
The timestamp in the tfdt refers to the first trun box and if there are
multiple trun boxes then the distance between the first timestamps will
grow.
At some point this distance reaches a threshold and triggers the
resetting of the first sample's timestamp of this trun box to be reset
to the tfdt.
This threshold is implemented for files where there is a jump in the
timeline between fragments and where this can be detected via a jump
between the end timestamp of the previous fragment and the tfdt of the
next. This behaviour is preserved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2414>
Hantro H1 and Rockchip VEPU2 drivers will pad the width/height to a
multiple of 16. In order to obtain the right JPEG size, the image needs
to be cropped using the S_SELECTION API. This support is added as best
effort since older drivers may emulate this by looking at the capture
queue width/height.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2392>
The gst_v4l2_object_set_crop() is used for removing buffer
alignment padding. Give it a name that better reflects
that usage. This helps to distinguish from cropping of the
input image (e.g. cropping at the image sensor on a captre
device), which can be unrelated to the memory buffer padding,
especially if scaling is involved.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2392>
mp4mux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.
By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2381>
flvmux can't negotiate caps with upstream/downstream and always outputs
specific caps based on the input streams. This will always happen before
it produces the first buffers.
By having the default aggregator negotiation enabled the same caps
would be pushed twice in the beginning, and again every time a
reconfigure event is received.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2381>
If a file includes a new version of a plugin that exits in the
registry, the output of gst-inspect is incorrect. The output has the
correct version but incorrect filename, and element description.
This seems to have also fixed some documentation issues.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2357>
In push mode (streaming), if the received chunk buffer size from _chain is bigger
than output buffer size, the flags of the divided-buffers are propagated to the
DISCONT flag from first received chunk buffer. This unexpected buffers contained DISCONT
flags are abnormally transformed when changing the sampling rate by audioresample element.
So unset unnecessary DISCONT flag before pad_push().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2322>
As this element is single threaded, we only need to stop the objects to
allow changing the format again. Fixes assertion notably on shutdown and
on some other situation where the format may be set twice without
actually activating the element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1991>
This is difficult to encounter in ordinary networks, but is
encountered when using tc-netem to add random delays to packets, and
also when your UDP stream is bonded over multiple links with varying
characteristics.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1979>
There is a chance that pool->buffers[index] sets BUFFER_STATE_QUEUED, but
it has not been queued yet which makes pool->buffers[index] still NULL.
At this time, if pool_streamff release all buffers with BUFFER_STATE_QUEUED
state regardless of whether the buffer is NULL or not, it will cause segfault.
To fix this, also check buffer when streamoff release buffer.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1948>
This patch fixes a seg.fault in gst_structure_new() with warnings as below.
GLib-GObject-WARNING **:
../gobject/gtype.c:4330: type id '0' is invalid
GLib-GObject-WARNING **:
can't peek value table for type '<invalid>' which is not currently referenced
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1920>
When doing only a single stream of audio/video this hardly matters,
but when doing many at the same time, the fact that you have to get
a hold of the glib global type-system lock every time you process a buffer,
means that there is a limit to how many streams you can process in
parallel.
Luckily the fix is very simple, by doing a cast rather than a full
type-check.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1890>
