Commit graph

825 commits

Author SHA1 Message Date
Tim Blechmann
810876d3d4 v4l2: silence valgrind warning
Valgrind complains about uninitialized memory used in an ioctl

    Syscall param ioctl(VKI_V4L2_G_TUNER).reserved points to uninitialised byte(s)
       at 0x719294F: ioctl (ioctl.c:36)
       by 0x3126A817: gst_v4l2_fill_lists (v4l2_calls.c:185)
       by 0x3126A817: gst_v4l2_open (v4l2_calls.c:589)
       by 0x3123F1C2: gst_v4l2_device_provider_probe_device (gstv4l2deviceprovider.c:122)
       by 0x3123F648: gst_v4l2_device_provider_device_from_udev (gstv4l2deviceprovider.c:301)
       by 0x3123F998: provider_thread (gstv4l2deviceprovider.c:395)
       by 0x796FA50: ??? (in /usr/lib/x86_64-linux-gnu/libglib-2.0.so.0.7200.4)
       by 0x710CAC2: start_thread (pthread_create.c:442)
       by 0x719DA03: clone (clone.S:100)
     Address 0x44008a34 is on thread 11's stack
     in frame #1, created by gst_v4l2_open (v4l2_calls.c:524)
     Uninitialised value was created by a stack allocation
       at 0x3126A024: gst_v4l2_open (v4l2_calls.c:524)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6760>
2024-04-29 15:42:09 +01:00
Tim Blechmann
e6480f6913 soup: fix thread name
thread names should be below 16char, otherwise they won't be shown on
linux.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6738>
2024-04-26 12:36:24 +01:00
Qian Hu (胡骞)
c3238be321 qtdemux: fix wrong full_range offset when parsing colr box
use colr_data[18] >> 7 to get full range information, instead
of colr_data[17] >> 7

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6641>
2024-04-15 07:52:29 +00:00
Jimmy Ohn
2f40a0c0d6 pulsedeviceprovider: Add is_default_device_name function and missing lock
Add is_default_device_name function to simplify compare device type
name and fix the missing lock when accessing default_sink_name and
default_source_name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6640>
2024-04-15 06:38:53 +00:00
Sebastian Dröge
6476fac04f rtpbin: Don't re-use a variable for a completely different purpose temporarily
During RTP-Info synchronization, clock_base was temporarily switched
from the actual clock-base to the base RTP time and then back some lines
later.

Instead directly work with the base RTP time. The comment about using a
signed variable for convenience doesn't make any sense because all
calculations done with the value are unsigned.

Similarly, rtp_clock_base was overridden with the rtp_delta when
calculating it, which was fine because it is not used anymore
afterwards. Instead, introduce a new variable `rtp_delta` to make this
calculation clearer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6639>
2024-04-15 01:27:28 +01:00
Sebastian Dröge
93f93847e8 rtpbin: Convert clock-base to extended RTP timestamp correctly
It's not in the same period as the current RTP base time but always in
the very first period. This avoids using it again at a much later time.

The code in question is only triggered with rtcp-sync=rtp-info.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6639>
2024-04-15 01:27:28 +01:00
Sebastian Dröge
d2b00b045a rtpjitterbuffer: Use an extended RTP timestamp for the clock-base
It is compared to other extended RTP timestamps all over rtpjitterbuffer
and since 4df3da3bab the initial extended RTP timestamp is not equal
anymore to the plain RTP time.

Continue passing a non-extended RTP timestamp via the `sync` signal for
backwards compatibility. It will always be a timestamp inside the first
extended timestamp period anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6639>
2024-04-15 01:27:27 +01:00
Sebastian Dröge
2b5930f6a0 wavpackparse: Use an unsigned integer for the block size calculations
It's never negative.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6564>
2024-04-08 09:34:13 +00:00
Sebastian Dröge
6953204a9c wavpackparse: Fix potential integer overflow on ID_ODD_SIZE blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6564>
2024-04-08 09:34:12 +00:00
Sebastian Dröge
18548cdd76 wavpackparse: Explicitly handle ID_WVX_NEW_BITSTREAM
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6564>
2024-04-08 09:34:12 +00:00
Elliot Chen
7e7f998f77 v4l2: fix error in calculating padding bottom for tile format
This is a regression while porting to arbitrary tile dimensions
introduced in !3424.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6565>
2024-04-08 02:22:36 +00:00
Tim-Philipp Müller
37bacb49a9 tests: rtpred: fix out-of-bound writes
Don't write more data to the buffer than we allocated
space for.

Fixes #3312

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6479>
2024-03-29 00:22:57 +00:00
Tim-Philipp Müller
3abc1c8c7b Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6408>
2024-03-20 00:07:00 +01:00
Tim-Philipp Müller
e49b86e82e Release 1.22.11 2024-03-19 22:01:08 +01:00
Haihua Hu
6a71f0e99d v4l2src: fix cannot reuse current caps when fixate caps in negotiation
when regotiation happens, v4l2src will check if it can reuse current caps,
but we need check if current caps is subset of all query caps from downstream
instead of check it with query caps one by one.

Assuming that the current caps is not the subset of first caps from query caps,
it will go to try fmt. when try fmt success, v4l2src will make pending_set_fmt
to TRUE and going to reset.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6393>
2024-03-18 15:30:01 +01:00
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
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
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
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
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
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
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
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
Jonas Kvinge
da03d2f2a0 meson: Set cpp_std to c++17 for TagLib
TagLib uses C++17 as of version 2.0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6097>
2024-02-12 11:55:33 +00:00
Tim-Philipp Müller
1a0df09c8f test: souphttpsrc: update test_icy_stream url
The old URL would result in multiple redirects, which
causes memory leaks with old versions of libsoup as
used on the CI.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6049>
2024-02-04 14:00:01 +01:00
Tim-Philipp Müller
600e105e64 Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5990>
2024-01-25 00:18:12 +00:00
Tim-Philipp Müller
3e41b8f18b Release 1.22.9 2024-01-24 18:21:13 +00:00
Dan Searles
19d3b14f51 rtspsrc: fix ttl setting for udpsink[1]
Fix ttl setting being incorrectly applied to udpsink[0] rather
than to udpsink[1].

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5967>
2024-01-23 23:52:25 +00:00
Dan Searles
ba5692005d rtspsrc: set multicast-iface on udpsinks
Copy rtspsrc property multicast-iface to its udpsinks to
allow messages over those sinks back to the server to work (and
prevent 'Network unreachable' warnings).

Closes: #3239
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5967>
2024-01-23 23:52:25 +00:00
Xavier Claessens
0ce3a2782c v4l2src: Consider framerate during caps selection
This simplifies the way it picks the closest caps to preference and take into
consideration the framerate to avoid picking high resolution at 5fps or so.
Simply calculate a "distance" of caps A and B from the preference and put
closest first, sorting by framerate first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5956>
2024-01-22 18:52:57 +00:00
Philippe Normand
36518d81e7 vpxdec: Use appropriate domain and code for decoding errors
STREAM domain and DECODE error is commonly used in other decoders. ENCODE is for
encoders.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5918>
2024-01-13 18:21:41 +00:00
Sanchayan Maity
daa60e39f9 rtphdrext-clientaudiolevel: Fix level value being written by the extension
When level value is greater than 127, it was being clamped but this clamped
value was not the one being actually used. For level values greater than 127
this resulted in an incorrect value being used. As an example, a level value
of 187, after and'ed with 0x7F, it would result in 0x3B being reported as the
level value.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5894>
2024-01-07 12:58:27 +00:00
Sebastian Dröge
a4dc820899 rtpsession: Only warn once if configured latency needs to be known but isn't yet
Otherwise we would warn about this once for every single packet until
the LATENCY event is received.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5874>
2023-12-29 12:46:23 +01:00
Chao Guo
0587c2754e v4l2object: clear old fds in poll when closing v4l2object
When reopening a v4l2 device, the v4l2object->poll will include some old fds,
which was assigned to this device before. If the pipeline opens multiple v4l2
devices, the old fd may been assigned to other v4l2 devices when reopening
devices.

This will cause the timing of the pipeline become confusing when polling devices,
leading functional abnormalities.

Therefore, when closing v4l2object, remove the old fds in poll to ensure that the
pipeline timing is normal.

Signed-off-by: Chao Guo <chao.guo@nxp.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5840>
2023-12-19 17:18:58 +00:00