Commit graph

4030 commits

Author SHA1 Message Date
Seungha Yang
6e9249b078 d3d11device: Fix adapter LUID comparison in wrapped device mode
Fix integer type mismatching

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3382
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6367>
2024-03-14 03:23:32 +00:00
Alexander Slobodeniuk
f04ea0c1be rtspsrc: remove 'deprecated' flag from the 'push-backchannel-sample' signal
It seems that it was added by accident when copying from push-backchannel-buffer

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6366>
2024-03-14 00:36:07 +00:00
Sebastian Dröge
eaffd61da6 mpg123audiodec: Correctly handle the case of clipping all decoded samples
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3365

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6360>
2024-03-13 17:13:51 +00:00
Tim-Philipp Müller
f04f86f3ee Revert "audiobasesink: Don't wait on gap events"
This reverts commit 8e923a8e2d.

This caused regressions, see #3303.

Without this commit, osxaudiosrc ! osxaudiosink won't work
right, but since that hasn't really been a huge problem
for years it's probably best to revert this until a proper
solution can be figured out.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6356>
2024-03-13 12:49:41 +00:00
Mikhail Rudenko
50033d03d3 rtsp-stream: clear sockets when leaving bin
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.

Fixes gstreamer/gst-rtsp-server#113

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6347>
2024-03-13 11:19:56 +00:00
Piotr Brzeziński
66283e8865 audiovisualizer: Don't wrap temporary memory in buffers
Avoids potentially ending up with the buffermemory pointing to already-freed or reused addresses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6349>
2024-03-13 07:19:39 +00:00
Piotr Brzeziński
f2ad031eff audioencoder: Avoid wrapping temporarily mapped memory with a GstBuffer and passing that to subclass
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>
2024-03-13 07:19:39 +00:00
Nirbheek Chauhan
fa387a3eb7 gsturi: Sort by feature name to break a feature rank tie
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>
2024-03-13 06:29:01 +00:00
Nirbheek Chauhan
acd40e7852 rtspsrc: Don't invoke close when stopping if we've started cleanup
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>
2024-03-12 19:22:47 +00:00
Seungha Yang
3bc068473b wasapi2: Fix task memory leak
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6218>
2024-03-08 22:11:31 +00:00
Seungha Yang
b793c3e03b wasapi: Fix alloc/free function mismatch
... and fix leak in wasapi device provider

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3326
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6218>
2024-03-08 22:11:31 +00:00
Seungha Yang
229d9cb4e1 asiosink: Fix channel selection
Fixing copy paste mistake

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3321
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6217>
2024-03-08 21:01:25 +00:00
Guillaume Desmottes
e62888c07f uridecodebin3: fix deadlock when switching input item
There was a race between urisourcebin src pad handlers.
One was starting the next item before the other was blocked.

See
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3297#note_2288799
for details.

Fix #3297

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6214>
2024-03-08 19:49:05 +00:00
Jan Schmidt
dea8b1cb37 identity: Don't refuse seeks unless single-segment=true
identity only needs to configure the internal seek segment if it's
aggregating upstream segments into 1. If it's not, don't break
other seek behaviour by refusing (for example) instant-rate change
seeks.

Fixes: #3363
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6307>
2024-03-08 18:41:52 +00:00
Mathieu Duponchelle
6f21f90747 onvif: tests: check for T flag on all packets
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6294>
2024-03-07 19:58:09 +00:00
Mathieu Duponchelle
8830b03ec1 rtpgstpay: flush on EOS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6294>
2024-03-07 19:58:09 +00:00
Mathieu Duponchelle
64bbeb106c rtponviftimestamp: make sure to set E and T bits on last buffer of lists
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6294>
2024-03-07 19:58:09 +00:00
Sebastian Dröge
01469a7de5 rtpgstpay: Delay pushing of event packets until the next buffer
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>
2024-03-07 19:58:09 +00:00
Elizabeth Figura
a1d1c74be1 qtdemux: Do not set channel-mask to zero
Leave it uninitialized, so that the downstream decoder will initialize it appropriately. Setting it to zero is wrong.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6288>
2024-03-07 13:34:54 +00:00
Jan Schmidt
375d16a9fa rtspsrc: Parse Speed/Scale before Range in responses
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>
2024-03-07 12:03:04 +00:00
Jan Schmidt
6a07ced605 rtspsrc: Handle queries and events with no manager
When doing direct output with no session manager, we still
want to respond to queries and events from downstream, so
install the handlers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6287>
2024-03-07 12:03:04 +00:00
Jan Schmidt
7ad4055557 rtspsrc: return NO_PREROLL on PLAYING->PAUSED too
When transitioning back to PAUSED and rtspsrc is live, return
NO_PREROLL so the pipeline knows to skip preroll here too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6287>
2024-03-07 12:03:04 +00:00
Jan Schmidt
ab6c205a8e rtponviftimestamp: Use gst_segment_to_stream_time_full()
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>
2024-03-07 12:03:04 +00:00
Jan Schmidt
df34adae9e gstsegment: Don't use g_return_val_if_fail()
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>
2024-03-07 01:35:09 +00:00
Jan Schmidt
1190761f7e dvbsubenc: Fix bottom field size calculation
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>
2024-03-07 00:29:12 +00:00
Piotr Brzeziński
adfefceea5 macos: Fix glimagesink not respecting preferred size
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>
2024-03-06 17:58:46 +00:00
Tim-Philipp Müller
dea9cfb5ee rtspsrc: Consider 503 Service Not Available when handling broken control urls
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6272>
2024-03-06 11:13:57 +00:00
Nirbheek Chauhan
6a18121fdc soup: Re-add soup-lookup-dep option
It's still useful on Linux since it ensures that the tests are going
to be built, since they use the same dep lookup as the plugin now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6198>
2024-03-04 14:58:26 +00:00
Arnaud Vrac
0db7773d5b adaptivedemux2: fix build with recent meson
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6198>
2024-03-04 14:58:26 +00:00
Nirbheek Chauhan
2b121be8f0 soup: Link to libsoup in all cases on non-Linux
We have unsolvable issues on macOS because of this, and the feature
was added specifically for issues that occur on Linux distros since
they ship both libsoup 2.4 and 3.0.

Everyone else should just pick one and use it, since you cannot mix
the two in a single process anyway.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1171

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6198>
2024-03-04 14:58:26 +00:00
Nirbheek Chauhan
2abbc2e0d9 good/tests: Don't enable soup tests if soup is disabled
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3268

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6198>
2024-03-04 14:58:26 +00:00
Olivier Crête
4f777ab6ae soup: Avoid using GUri before GLib 2.66
Let's use gpointer for now

Fixes: #3169
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6198>
2024-03-04 14:58:26 +00:00
Olivier Crête
b4003f4449 adaptivedemux2: Parse cookies in downloadhelper
We need to parse any cookie headers, otherwise we end up
sending back attributes likes "Secure" and "httponly" which break
some servers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6198>
2024-03-04 14:58:26 +00:00
Edward Hervey
30738b09c1 plugins: Fix wrong enum usage
gcc 13 now detects conflicting enum usages. Fix the various cases where it was wrong

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6234>
2024-02-28 01:18:22 +00:00
Seungha Yang
0640147cee jpegdec: Fix progressive/interlaced detection
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>
2024-02-27 23:58:15 +00:00
Seungha Yang
42c07de96c jpegdec: Remove trailing white space
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6229>
2024-02-27 23:58:15 +00:00
Nirbheek Chauhan
1e384e5414 rtspsrc: Increase rank to PRIMARY for autoplug purposes
This affects autoplug by gst_element_make_from_uri() in, for example,
uridecodebin. The element should've already been PRIMARY rank, but it
was NONE because gst_element_make_from_uri() doesn't ignore NONE rank
elements when searching for element factories, unlike decodebin.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/502

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6231>
2024-02-27 17:23:43 +00:00
Alexander Slobodeniuk
00102b6c16 d3d11window_win32: fix crash on RC unprepare() vs window_proc()
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>
2024-02-27 15:41:41 +00:00
Seungha Yang
7ff7ced388 wasapi2: Respect ringbuffer buffer/latency time
Decide buffer size based on configured buffer/latency time
if low-latency is disabled, so that ringbuffer can buffer more
than minimum required size.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/2870
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6215>
2024-02-24 12:46:31 +00:00
Jan Schmidt
fb8131b7da rtspsrc: Reset combined flows after a seek before restarting
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>
2024-02-24 11:20:51 +00:00
Víctor Manuel Jáquez Leal
edd2b3c6ee vabasedec: disable derived images for i965 driver
Since it has a very poor performance at reading derived images, which is the
most common use case for decoders.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6204>
2024-02-23 12:23:09 +01:00
Víctor Manuel Jáquez Leal
6a42ed20ec vaallocator: remove runtime mapping selection
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>
2024-02-23 12:23:09 +01:00
Seungha Yang
e376d06845 wasapi2: Fix choppy rendering
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>
2024-02-19 10:21:58 +00:00
Robert Mader
2f1b123f7f v4l2codecs: h264: Fix a memory leak on renegotiation
We only use this anchor when streaming, in which case output_state is
set and needs to get unreffed.
This is in line with how it is handled for all other codecs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6134>
2024-02-16 23:41:58 +00:00
Loïc Molinari
44faeb532d video: Fix NV12_16L32S video frame size
The size of a NV12_16L32S video frame is bigger than expected because
it uses the size of a Y tile to compute the interleaved UV plane
size. Get the right UV tile size instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6135>
2024-02-16 18:13:32 +00:00
Edward Hervey
4e01c01483 urisourcebin: Don't acquire STATE_LOCK if shutting down
If we are shutting down (PAUSED->READY) we shouldn't take the STATE LOCK since
this function is being called from a streaming thread (which is trying to be
deactivated while the STATE LOCK is held)

Fixes #3292

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6123>
2024-02-15 11:11:07 +00:00
Tim-Philipp Müller
dcd9d8a87d Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6111>
2024-02-13 16:27:38 +00:00
Tim-Philipp Müller
29d6413c3f Release 1.22.10 2024-02-13 14:39:08 +00:00
Piotr Brzeziński
6e59282af2 macos: Fix gst_macos_main() terminating whole process before returning a value
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>
2024-02-12 18:25:34 +01:00
Piotr Brzeziński
2f4e8d14cf macos: Set activation policy in osxvideosink and glimagesink
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>
2024-02-12 18:25:18 +01:00