Remove our own translation in the mouse event capture code, as that
translation will be done through the navigation interface.
Tested by resizing the window created by:
gst-launch-1.0 -v videotestsrc ! navigationtest ! glupload ! glcolorconvert ! tee name=t ! gtkglsink
and checking that the cursor follows the mouse as expected.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/974>
qtmux attempts to choose between writing a 32-bit stco chunk offset table
when it can, but switch to a 64-bit co64 table when file offsets go over
4GB.
This patch fixes a problem where the atom handling code was checking
mdat-relative offsets instead of the final file offset (computed by
adding the mdat position plus the mdat-relative offset) - leading to
problems where files with a size between 4GB and 4GB+offset-of-the-mdat
would write incorrect STCO tables with some samples having truncated
32-bit offsets.
Smaller files write STCO correctly, larger files would switch to
co64 and also output correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/970>
skip_try_fmt_probes is set to TRUE for v4l2src to skip interlace-mode and
colorimetry when probe caps. gst_v4l2_object_set_format_full() will add
colorimetry back to caps when iterating over the negotiated caps. There is
one case that v4l2src is first in preview state then starts recording.
v4l2src caps will change with an additional interlace-mode structure after
renegotiation, then v4l2src needs to reset. But this camera driver can't
orphan buffer pool, it causes require buffer failed as streaming is still
in active state.
To fix this, also need to add interlace-mode back to caps for camera to
avoid reset.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/961>
* Try to make variable and function names more clear.
* Add plenty of comments describing the logic step-by-step.
* Improve the logging around this, making the logs easier to read and
understand when debugging these issues.
* Revise the logic of packets that are actually beyond saving in doing
the following:
1. Do an optimistic estimation of which packets can still arrive.
2. Based on this, find which packets (and duration) are now hopelessly
lost.
3. Issue an immediate lost-event for the hopelessly lost and then add
lost/rtx timers for the ones we still hope to save, meaning that if
they are to arrive, they will not be discarded.
* Revise the use of rtx-delay:
Earlier the rtx-delay would vary, depending on the pts of the latest
packet and the estimated pts of the packet it being issued a RTX for,
but now that we aim to estimate the PTS of the missing packet accurately,
the RTX delay should remain the same for all packets.
Meaning: If the packet have a PTS of X, the delay in asked for a RTX
for this packet is always a constant X + delay, not a variable one.
* Finally ensure that the chaotic "check-for-stall" tests uses timestamps
that starts from 0 to make them easier to debug.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/952>
GstGLVideoItem is required to render input video in the overlay's qml.
And currently qmlgloverlay requires to set this GstGLVideoItem to its widget property.
Instead of fetching GstGLVideoItem from the overlay's root object (root-item prop),
and setting it back as a widget (widget prop),
proposing to use found GstGLVideoItem in the current object hierarchy (passed in qml-scene) by default.
Also useful in Python, which solves the issue when casting gpointer <=> QQuickItem* is required.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/919>
Would spew warnings on the rpi4 when calling into
gst_v4l2_object_get_codec_caps() from the probe_and_register()
function since the v4l2_debug category initialisation would
only be done later as part of the element/device provider
registration.
Also log things in the probe function to the v4l2 category
instead of the default category while we're at it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/956>
Seek events are sent upstream on each sink, so if we receive multiple
seeks with the same seqnum, we must only perform one seek, not N seeks
where N = the number of sinks in the pipeline connected to rtspsrc.
This is the same thing done by demuxers like qtdemux or matrsokademux.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/938>
This code can be fragile, since it is very exacting in the timestamps
that it will accept. Add more logging so it's easier to debug issues
and figure out whether it's a bug in the calculation or something
wrong in the incoming buffers.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/934>
gst_rtp_sbc_pay_flush_buffers() is a misleading name. A better name would
be gst_rtp_sbc_pay_drain_buffers(), because that's what it does, it drains
any leftover queued data and pushes it downstream. "Flushing" in GStreamer
typically means to throw away any queued data and not process/push it
downstream.
Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>
GstAdapter must be flushed in some cases (flush, new segment, state change)
Without it, it may, for example, push some leftover buffer from old
segment in new segment. This, in general, breaks timestamps.
See GstAdapter documentation for more.
Signed-off-by: Michal Dzik <michal.dzik@streamunlimited.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/merge_requests/700>
