Sebastian Dröge
4b6cbca300
flac: Add wrap file and add fallback for it to the flac plugin
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6560 >
2024-04-07 13:57:40 +00:00
Tim Blechmann
88dd91cb30
v4l2: enforce a pixel aspect ratio of 1/1 if no data are available
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6559 >
2024-04-07 14:17:19 +01:00
Philipp Zabel
993d8b89bc
v4l2: allocator: Fix unref log/trace on memory release
...
Use gst_object_unref() instead of g_object_unref() in
gst_v4l2_allocator_release(), so refcounting log and
tracer get to know about this unref.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6556 >
2024-04-06 23:41:52 +00:00
Elliot Chen
7aa75e590a
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/6554 >
2024-04-06 19:10:25 +01:00
Sebastian Dröge
984b1f413a
wavpackparse: Use an unsigned integer for the block size calculations
...
It's never negative.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6541 >
2024-04-04 17:33:48 +01:00
Sebastian Dröge
078ef786d2
wavpackparse: Fix potential integer overflow on ID_ODD_SIZE blocks
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6541 >
2024-04-04 17:33:48 +01:00
Sebastian Dröge
99bdbd78ca
wavpackparse: Explicitly handle ID_WVX_NEW_BITSTREAM
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6541 >
2024-04-04 17:33:48 +01:00
Tim-Philipp Müller
d2aeaeb73f
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/6478 >
2024-03-29 00:22:16 +00:00
Haihua Hu
bb43b96d2c
v4l2src: need maintain the caps order in caps compare when fixate
...
if the calculated "distance" of caps A and B from the preference are
equal, need to keep the original order instead of swap them
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6473 >
2024-03-28 20:05:27 +00:00
Jan Schmidt
2980981618
rtpmp4adepay: Set duration on outgoing buffers
...
If we can calculate timestamps for buffers, then set the duration
on outgoing buffers based on the number of samples depayloaded.
This can fix the muxing to mp4, where otherwise the last packet
in a muxed file will have 0 duration in the mp4 file.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6456 >
2024-03-27 19:48:43 +00:00
Hou Qi
c0d35575a8
v4l2: Also set max_width/max_height if enum framesize fail
...
Some driver doesn't implement enum_framesize. The maximum supported
size can be got by trying format with a very large size. Also need
to set max_width/max_height for this case, otherwise default encoded
buffer size 256kB is too small.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6430 >
2024-03-22 17:27:58 +00:00
Tim-Philipp Müller
175d116f94
Back to development
2024-03-22 01:38:06 +01:00
Tim-Philipp Müller
0d0a1d9d16
Release 1.24.1
2024-03-21 21:47:53 +01:00
Edward Hervey
db6803bd55
adaptivedemux2: Don't use g_str_equal on potentially NULL strings
...
It is only meant to be used as a callback. The fallback macro uses strcmp which
doesn't handle NULL strings gracefully. Instead use g_strcmp0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6410 >
2024-03-20 10:49:02 +01:00
Edward Hervey
77fa0ae0e7
hlsdemux2: Avoid NULL pointer usage
...
The pending/current variant are both NULL when the demuxer is resetted.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6410 >
2024-03-20 10:49:02 +01:00
Edward Hervey
9ce063f5f6
adaptivedemux2: Handle context going away
...
This issue can happen when the scheduler loop was stopped (and context went
away). We no longer want to push/pop main context threads.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6410 >
2024-03-20 10:49:02 +01:00
Edward Hervey
257de579b3
hlsdemux2: Improve detection of playlist updates
...
In the case we are not updating an existing playlist, we only want to reset the
download error count if the URI we are downloading was not the previous one we
were trying to load
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6410 >
2024-03-20 10:49:02 +01:00
Alexander Slobodeniuk
6e57362f35
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/6363 >
2024-03-13 21:14:27 +00:00
Sebastian Dröge
5a2f99a255
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/6357 >
2024-03-13 14:23:56 +00:00
Piotr Brzeziński
6b369d8470
qtdemux: Fix wrapping temporary memory in buffers
...
That memory can disappear at any moment, doesn't cost much to just copy those few bytes.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6339 >
2024-03-13 12:36:28 +00:00
Nirbheek Chauhan
bcb016d6d2
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/6330 >
2024-03-11 18:32:12 +00:00
Michael Tretter
555bb8ece2
meson: Fix description in qt options
...
The qt-x11 description contains a copy/paste error from the qt-wayland option.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6313 >
2024-03-10 12:24:21 +00:00
Mathieu Duponchelle
172221a2cf
rtpgstpay: flush on EOS
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6298 >
2024-03-08 03:47:38 +00:00
Sebastian Dröge
d804e133e0
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/6298 >
2024-03-08 03:47:38 +00:00
Elizabeth Figura
a4d3d80e95
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/6296 >
2024-03-08 02:18:53 +00:00
Jan Schmidt
538cafbd9c
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/6295 >
2024-03-08 00:51:50 +00:00
Jan Schmidt
464cd9f9a3
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/6295 >
2024-03-08 00:51:50 +00:00
Jan Schmidt
fc3be23863
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/6295 >
2024-03-08 00:51:50 +00:00
Tim-Philipp Müller
0f3099ef5c
rtspsrc: Consider 503 Service Not Available when handling broken control urls
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6267 >
2024-03-06 01:35:01 +00:00
Tim-Philipp Müller
2c7bb61580
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6264 >
2024-03-05 14:00:10 +00:00
Tim-Philipp Müller
b125253cad
Release 1.24.0
2024-03-04 23:59:25 +00:00
Nirbheek Chauhan
cf2238a522
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/6226 >
2024-02-27 11:36:01 +00:00
Edward Hervey
a3980f4838
docs: Use Discourse and Matrix as prefered communication channels
...
Part of: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6220
2024-02-27 09:35:47 +01:00
Seungha Yang
125c89319a
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/6181 >
2024-02-26 23:21:44 +09:00
Seungha Yang
3afeb73538
jpegdec: Remove trailing white space
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6181 >
2024-02-26 23:14:54 +09:00
Tim-Philipp Müller
d474de8ff0
Release 1.23.90
2024-02-23 18:20:11 +00:00
Nirbheek Chauhan
4fc56a08ee
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/6197 >
2024-02-23 11:47:47 +05:30
Matthew Waters
697b35fe58
examples/qmlsinnk-multisink: allow running with leaks tracer
...
Include a gst_deinit() after the qml engine has been destroyed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6160 >
2024-02-22 10:26:39 +00:00
Matthew Waters
f1637a3601
examples/qml: fix some leaks in the multisink example
...
A GstPad was being leaked and possibly the qmlglsink element depending
on if Qt runs the scenegraph thread again when destroying the example
video item.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6160 >
2024-02-22 10:26:39 +00:00
Matthew Waters
392fd00f4c
qml, qml6: Fix leak of QSGMaterial/Geometry (and therefore a possible GstBuffer)
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6160 >
2024-02-22 10:26:31 +00:00
Matthew Waters
2dae3775d9
qml6: fix a leak of the wrapped QSGTextures
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6160 >
2024-02-22 10:24:24 +00:00
Sebastian Dröge
69e4564c87
rtphdrext-clientaudiolevel: Fix typo in documentation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6175 >
2024-02-21 17:25:43 +00:00
Arnaud Vrac
9e2e456d9f
adaptivedemux2: fix build with recent meson
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6168 >
2024-02-21 13:53:40 +00:00
Tim-Philipp Müller
0a6948ee20
rtppassthroughpay: fix critical in gst-inspect
...
gst_segment_to_running_time() will fail noisily
if the segment has not been initialised yet.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6151 >
2024-02-21 11:25:10 +00:00
Nirbheek Chauhan
11f6984bf5
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/6156 >
2024-02-21 09:27:59 +05:30
Jan Schmidt
f7e494f348
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/6137 >
2024-02-21 01:50:13 +00:00
Maksym Khomenko
ccf544a50e
osxaudio: add mapping for top/left/right surround channels
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5731 >
2024-02-20 08:03:15 +00:00
Maksym Khomenko
f1e02ebb92
osxaudio: correct mapping for left/right surround
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5731 >
2024-02-20 08:03:15 +00:00
Marc Leeman
eb17de27d6
qt6: search in /usr/lib/qt6/bin/ for qsb
...
In Debian and possibly other distributions, qsb (qt6-shader-baker) is
not in the default path, but in a QT6 specific path. Search there too
Applied changes from Nirbheek
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6095 >
2024-02-19 12:29:32 +00:00
Jochen Henneberg
6608b89977
rtpxqtdepay: Enabled header extension aggregation
...
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
5d1d0cf9a5
rtpmp4gdepay: Enabled header extension aggregation
...
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
75849c63c8
rtpsbcdepay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
3fffcd021a
rtpvorbisdepay: Enabled header extension aggregation
...
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
e1e7421982
rtpmp4vdepay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
334ceaca21
rtptheoradepay: Enabled header extension aggregation
...
Because this depayloader may build several output buffers within one
process run we push them all into a GstBufferList and push them out at
once to make sure that each buffer gets notified about each header
extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
0a4918a509
rtpsv3vdepay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
d810049f01
rtpmp4adepay: Enabled header extension aggregation
...
Because this depayloader may build several output buffers within one process
run we push them all into a GstBufferList and push them out at once to
make sure that each buffer gets notified about each header extension.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
90b5d2eb93
rtpklvdepay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:17 +00:00
Jochen Henneberg
2c3f169ebb
rtpjpegdepay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:16 +00:00
Jochen Henneberg
460813f7ee
rtpj2kdepay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:16 +00:00
Jochen Henneberg
ae3a00abd2
rtph263pdepay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:16 +00:00
Jochen Henneberg
4fd4c240e0
rtph263depay: Enabled header extensions aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:16 +00:00
Jochen Henneberg
ae5bdaa7e1
rtph261depay: Enabled header extension aggregation
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5378 >
2024-02-19 11:23:16 +00:00
Priit Laes
4e782da32e
cacasink: add driver selection support from the pipeline
...
https://bugzilla.gnome.org/show_bug.cgi?id=599018
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5491 >
2024-02-19 07:50:15 +00:00
Tim-Philipp Müller
88412ef100
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6126 >
2024-02-15 16:38:53 +00:00
Tim-Philipp Müller
88751d4110
Release 1.23.2
2024-02-15 15:37:17 +00:00
Sebastian Dröge
499474a76d
Revert "rtpvp8pay: Use GstBitReader instead of dboolhuff implementation from libvpx"
...
This reverts commit b730e7a1b2
.
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3300
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6116 >
2024-02-14 15:45:24 +00:00
Mathieu Duponchelle
91317aacaf
webrtcbin, rtpbin: check before setting properties on jitterbuffer
...
In rtpbin we already systematically check for all property names
except latency, correct that.
In webrtcbin we need to check before trying to use the do-retransmission
property.
This is useful for the case where an element like identity gets passed
to rtpbin's request-jitterbuffer property, when the application wants
to use webrtcbin in an SFU situation, with no reordering and no added
latency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6112 >
2024-02-14 08:52:50 +00:00
Sebastian Dröge
c726add352
rtpfunnel: Handle NTP-64 RTP header extension in caps similar to TWCC
...
This is another header extension that is handled by rtpsession and needs
to be preserved in the caps that are created by rtpfunnel.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6109 >
2024-02-14 08:05:33 +00:00
Tim-Philipp Müller
b87093207c
gst-plugins-good: update translations
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6113 >
2024-02-14 04:43:00 +00:00
Sebastian Dröge
17e7af7181
rtpfunnel: Also write TWCC RTP header extension into buffer list buffers
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6110 >
2024-02-14 01:56:20 +00:00
Philippe Normand
6f778eebf9
dashdemux2: Basic support for container-specific-track-id tag
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6041 >
2024-02-12 10:37:29 +00:00
Philippe Normand
e9ecde83a7
matroska-demux: Basic support for container-specific-track-id tag
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6041 >
2024-02-12 10:37:29 +00:00
Philippe Normand
30bb88a91b
qtdemux: Basic support for container-specific-track-id tag
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6041 >
2024-02-12 10:37:29 +00:00
Nirbheek Chauhan
d0ae93771e
meson: Don't use fs.copyfile() for qt6 resources
...
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3285
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6077 >
2024-02-09 11:14:09 +05:30
Ignazio Pillai
34741e1db2
cutter: add audio-level-meta
...
Set GstAudioLevelMeta on buffers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5771 >
2024-02-08 13:52:40 +00:00
Nirbheek Chauhan
f6f448bb80
meson: Fix several warnings in the build
...
Deprecations, incorrect options, etc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6058 >
2024-02-06 23:57:17 +00:00
Nirbheek Chauhan
63322705c8
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/6058 >
2024-02-06 23:57:17 +00:00
Tim-Philipp Müller
2111d6f015
Back to development
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6066 >
2024-02-06 18:29:31 +00:00
Tim-Philipp Müller
9255e397f0
Release 1.23.1
2024-02-06 16:43:27 +00:00
Tim-Philipp Müller
e7d771903e
meson_options.txt: fix meson warning about default bool values being a string
2024-02-06 16:37:13 +00:00
Nirbheek Chauhan
a5cb2ef9cd
meson: Print a useful error message when qt windowing is not found
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6040 >
2024-02-03 05:42:26 +00:00
Hou Qi
aa68b5e02a
hlsdemux2/m3u8: use GstClockTimeDiff to do timestamp comparison
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5234 >
2024-02-01 08:41:23 +00:00
Sebastian Dröge
b730e7a1b2
rtpvp8pay: Use GstBitReader instead of dboolhuff implementation from libvpx
...
All compressed frame header values that are read as part of the
payloader are encoded as bits with 50:50 probability, and as such are
just the plain bits as they are.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5810 >
2024-01-31 16:52:28 +00:00
Daniel Morin
0a55c86e6a
rtspsrc: update rtsp url on redirect
...
- If a redirect took place on a GET when rtsp is tunneled we update the
rtsp url too.
- log source and final destination on redirect
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5222 >
2024-01-31 11:43:45 +00:00
Thibault Saunier
e1a8ce16b4
matroskademux: Lower verbosity of some often happenning warnings
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6011 >
2024-01-30 09:09:22 +00:00
Thibault Saunier
77e7efe407
qtdemux: Lower verbosity of some often happenning warnings
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6011 >
2024-01-30 09:09:22 +00:00
Tim-Philipp Müller
c84285d44d
meson: bump Meson requirement to >= 1.1 for all modules
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6002 >
2024-01-29 01:11:55 +00:00
Jonas Kvinge
a35723d531
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/5995 >
2024-01-27 12:44:48 +00:00
Mathieu Duponchelle
03d07e8e52
vpxenc: fix warning about decreasing PTS on first frame
...
The fields used to track this state should be initialized when
codec->inited is FALSE on set_format, not TRUE
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3200
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5876 >
2024-01-25 08:17:26 +00:00
Jonas K Danielsson
b0becfa46b
splitmuxsrc: Use natural ordering to find files
...
Today when using the `splitmuxsrc` on a collection of files named as:
```
item0.mkv
item1.mkv
item2.mkv
[...]
item10.mkv
item11.mkv
[...]
```
You will get a continuous stream made in the order of:
```
item0.mkv -> item1.mkv -> item10.mkv -> item11.mkv -> [...]
```
You can fix this by having smarter names of the items:
```
item000.mkv
item001.mkv
item002.mkv
[...]
item010.mkv
item011.mkv
[...]
```
Will get you:
```
item000.mkv -> item001.mkv -> item003.mkv -> item004.mkv -> [...]
```
But, we could also "fix" the former case by using natural ordering when
comparing the files in gstsplitutils.c.
Fixes #2523
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4491 >
2024-01-24 20:15:19 +00:00
Dan Searles
1d02d7eda0
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/5955 >
2024-01-23 13:54:51 +00:00
Dan Searles
da55b953a1
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/5955 >
2024-01-23 13:54:51 +00:00
Guillaume Desmottes
fae6fbaa6b
flvdemux: don't re-use segment from one stream if the other has buffer earlier
...
Fix first audio buffers being out of segment because the audio stream
is starting earlier than the video one which was the first demuxed.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5940 >
2024-01-19 11:05:05 +01:00
Guillaume Desmottes
632ee523fb
flvdemux: factor out ensure_new_segment()
...
- Use the pad instead of the element for logs, so it's clearer on which
pad this segment will be pushed.
- One copy was checking for invalid seq num, the other was not.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5940 >
2024-01-19 11:05:01 +01:00
Hou Qi
2539bb0b1d
rtpjitterbuffer: Fix build warning in rtp_jitter_buffer_append_query()
...
This is to fix build warnings when using [-Wmaybe-uninitialized]
../gst/rtpmanager/rtpjitterbuffer.c:1237:10: warning: 'head' may be used uninitialized [-Wmaybe-uninitialized]
1237 | return head;
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5907 >
2024-01-13 15:00:19 +00:00
Philippe Normand
8a99589d2c
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/5916 >
2024-01-12 14:10:36 +00:00
Olivier Crête
814f21557f
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/5906 >
2024-01-11 18:06:59 +00:00
Sebastian Dröge
6fa41f78bb
rtpsession: Remove some unused fields
...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5899 >
2024-01-08 12:57:04 +02:00
Sanchayan Maity
00bbac6541
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/5893 >
2024-01-07 16:00:18 +05:30