Since commit 4d86f994, when setting an RTSP media both shared and
reusable, streaming cannot be restarted after the first time all the
clients disconnect. That happens because the sockets (unlike
addresses) of GstRTSPStream are not cleared in
gst_rtsp_stream_leave_bin, and on restart sockets and addresses are
not allocated in gst_rtsp_stream_allocate_udp_sockets, and then the
check in create_sender_part fails. Fix this by clearing sockets in
gst_rtsp_stream_leave_bin.
Fixesgstreamer/gst-rtsp-server#113
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6347>
Memory from gst_adapter_map() could live shorter than the GstMemory that the GstBuffer wraps around it, which in lucky
cases 'just' caused a re-use of the same memory for multiple (potentially still in use!) input buffers, but could easily
end up pointing to an already-freed memory.
Manifested when an AudioToolbox encoder kept getting silence inserted in seemingly random circumstances, turned out
to be the memory being re-used by GStreamer at the same time that the AT API was processing it...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6349>
This matches autoplug in other places such as decodebin, otherwise we
will pick "randomly" based on the order in which plugins are
registered, which is mostly dependent on the order in which readdir()
returns items.
So let's make it predictable.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6348>
When we're doing a state change from PLAYING to NULL, first we invoke
gst_rtspsrc_loop_send_cmd_and_wait (..., CMD_CLOSE, ...) during
PAUSED_TO_READY which will schedule a TEARDOWN to happen async on the
task thread.
The task thread will call gst_rtspsrc_close(), which will send the
TEARDOWN and once it's complete, it will call gst_rtspsrc_cleanup()
without taking any locks, which frees src->streams.
At the same time however, the state change in the app thread will
progress further and in READY_TO_NULL it will call gst_rtspsrc_stop()
which calls gst_rtspsrc_close() a second time, which accesses
src->streams (without a lock again), which leads to simultaneous
access of src->streams, and a segfault.
So the state change and the cleanup are racing, but they almost always
complete sequentially. Either the cleanup sets src->streams to NULL or
_stop() completes first. Very rarely, _stop() can start while
src->streams is being freed in a for loop. That causes the segfault.
This is unlocked access is unfixable with more locking, it just leads
to deadlocks. This pattern has been observed in rtspsrc a lot: state
changes and cleanup in the element are unfixably racy, and that
foundational issue is being addressed separately via a rewrite.
The bandage fix here is to prevent gst_rtspsrc_stop() from accessing
src->streams after it has already been freed by setting src->state to
INVALID.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6346>
And also re-timestamp them with the current buffer's PTS.
Not doing so keeps the timestamps of event packets as
GST_CLOCK_TIME_NONE or the timestamp of the previous buffer, both of
which are bogus.
Making sure that (especially) the first packet has a valid timestamp
allows putting e.g. the NTP timestamp RTP header extension on it.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6294>
Parse the speed and scale in the server's response
*before* the range, so that the range start/stop
are swapped (or not swapped) correctly based
on the server's actual chosen values. Otherwise,
the old rate from the segment is used - what the
last seek asked for, but not necessarily what
the server chooses.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6287>
In the situation where playback starts from a keyframe before
the target playback segment, then the first buffers will be
outside the configured segment and gst_segment_to_stream_time()
will return GST_CLOCK_TIME_NONE unconditionally.
If drop-out-of-segment is false, the RTP buffers will not be
dropped, but will be sent witout ONVIF extension timestamps
and given GST_CLOCK_TIME_NONE timestamps on the receiver.
Instead, use gst_segment_to_stream_time_full() to extrapolate
stream time outside the segment so that such buffers still
get assigned their correct timestamps on the receiver.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6287>
Don't use g_return_val_if_fail() to catch the
open-ended segment or empty segment cases in
gst_segment_to_running_time_full()
g_return_val_if_fail() is for programmer errors,
and can be compiled out with a flag.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6279>
Don't accidentally include the stuffing byte (if present)
into the bottom field size. It should only be included in the
total segment length.
Fixes problems with FFmpeg not rendering the subtitles
with a stuffing byte, giving a "Invalid object location!" error.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6281>
Cocoa version of glwindow only checks the preferred size upon window creation. glimagesink sets the size right before
calling gst_gl_window_show(), which might be way after the window is created in some cases. If the size was set too
late, glimagesink on macOS would remain 320x240 unless manually resized.
This change makes sure to resize the existing window when _show() is called.
Curiously, this has always been an issue, but went from manifesting every once in a while to being almost completely
broken once old event loop workarounds were removed and gst_macos_main() was introduced.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6280>
If input height and parsed one are identical, do not consider it as interlaced
Fixing below pipeline:
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420,width=640,height=10 \
! jpegenc ! jpegparse ! jpegdec ! videoconvert ! autovideosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6229>
Unprepare method posts WM_GST_D3D11_DESTROY_INTERNAL_WINDOW
command to the window queue, and from that moment considers
internal_hwnd to be released, and so it sets it to null.
The problem is that it's possible that right at that moment
the window thread might be already processing some other
command, or just another command might be already in the queue.
On practice we met a crash when WM_PAINT got processed in between
(unprepare already finished and WM_GST_D3D11_DESTROY_INTERNAL_WINDOW
was not handled yet)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6230>
After a flushing seek, rtspsrc doesn't reset the last_ret value for
streams, so might immediately shut down again when it resumes pushing
buffers to pads due to a cached `GST_FLOW_FLUSHING` result
Prevent a stored flushing value from immediately stopping
playback again by resetting pad flows before (re)starting
playback.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6216>
The original idea was to select the type of mapping (either using derive images
or downloading the image) in runtime, under the assumption that both methods
shared the same memory layout (offsets and strides), because a single
GstVideoMeta is assigned by the buffer pool at allocation time. Nonetheless, in
recent hardware this assumption is invalid, raising memory access errors.
This patch removes completely the mapping type selection at runtime, using the
method selected when the allocator is configured, synced with the bufferpool
allocation.
This problem was fixed originally for iHD driver only. But now it makes sense to
remove all of it.
Original-patch-by: Mengkejiergeli Ba <mengkejiergeli.ba@intel.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6204>
This reverts questionable commit 009bc15f33
which looks completely wrong.
The GstWasapi2RingBuffer:buffer_size variable is used to
calculate available buffer size we can write
(i.e., available size = buffer_size - padding_size).
But the commit makes the size to be exactly same as buffer period.
Then, it can confuse this element as if the endpoint buffer is full on
I/O event callback (if padding size is equal to buffer period)
but it's not true.
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2870
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6140>
Removes the usage of [NSApp terminate] to avoid killing the process and thus never actually returning a value.
The new way is just to use [NSApp stop] and send an event, since stop only happens after an event is processed.
Unlike terminate, stop will only halt the event loop, not the whole process.
This uses an NSApplicationDelegate to listen for NSApp finishing the launch process, and then signals the 'main' thread
to proceed. That makes sure to never call [NSApp stop] before NSApp is actually running, which could happen if the
provided 'main' function finished quickly enough.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6103>
Upon creating a window, glimagesink and osxvideosink now set the policy to
NSApplicationActivationPolicyRegular, which lets us show an icon in the Dock
for convenience and appear in the top menu bar like other apps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6103>
Setting the policy to NSApplicationActivationPolicyAccessory by default makes
sure that we can activate windows programmatically or by clicking on them.
Without that, windows would disappear if you clicked outside them and there
would be no way to bring them to front again. This change also allows osxvideosink
to receive navigation events correctly.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6103>
WebKit commit b12e7ed2ad3a ("[WPE] Upstream the new WPE platform API
https://bugs.webkit.org/show_bug.cgi?id=265286")[1] added a `WPEView` typedef
which clashes with our `WPEView` class.
Rename the `WPEView` class to `GstWPEThreadedView` to avoid the collision.
Also prefix the `WPEContextThread` class with `Gst` and rename the
source files to reflect the new class name and use lowercase while at it
for consistency
[1] b12e7ed2ad
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6096>
