Commit graph

1415 commits

Author SHA1 Message Date
Jan Schmidt
c1a1584dde splitmuxsrc: Don't create part reader elements initially
Only create the part reader elements internally the first time
the part is activated. Saves some startup time when preloading
a large number of fragments

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
8a1fab9594 splitmuxsrc: Drop lock when unpreparing parts
Parts may emit bus messages that want to take the splitmuxsrc
lock and prevent the downward state change. Avoid a deadlock
after a part sends an error message by taking a ref and
dropping the lock around the unprepare call

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
ec1c6c5b60 splitmuxsrc: Make sure to re-take lock
In the error path when activating a part fails, make
sure to re-take the splitmuxsrc lock before returning
to the caller.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
44005ab9fb splitmuxsink: Fix race in unit tests. Add fragment-id to messages
Publish fragment-id in the messages that splitmuxsink and splitmuxsrc
send, so when they are received out of order (due to async finalization,
for example), they can still be identified / ordered correctly.

Fix a race in the splitmuxsink unit test where messages might be
received out of order

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
356710f6fa splitmuxsrc: Document new properties and signals
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
64fd2b265f splitmuxsrc: Add num-lookahead property
Add a `num-lookahead` property that will 'prepare' a number of
fragments in advance of the playhead if they have been deactivated
or closed by a limited number of `num-open-fragments`. It can help
to avoid any play stalls reading the indexes or headers of the next
file from high-latency media or on resource limited machines.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
93c04e7473 splitmuxsrc: Rename some internal terminology
A part reader can be 'loaded' (prepared, but not currently outputting anything)
or 'playing' (actively being used to output data)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
3121eeeb08 splitmuxsrc: Allow adding fragments during playback
Trigger measurement / inclusion of new fragments into
the playback timeline if they are added after the
element is already running.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:12 +10:00
Jan Schmidt
ed03e8f8ab splitmuxsink: Add fragment offset and duration to message
Publish the playback offset for and duration into the
splitmuxsink-fragment-closed bus message as each fragment
finishes.

These can be passed to splitmuxsrc via the 'add-fragment'
signal to avoid splitmuxsrc measuring all files on startup

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:05 +10:00
Jan Schmidt
b0df6ee408 splitmuxsink: Fix a race in fragment switching with async handling
Only do output/muxer operations at the output side of splitmuxsink
to avoid races if fragments are small, by moving the RUNNING_TIME
qdata setting.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:05 +10:00
Jan Schmidt
eca97e7940 splitmuxsink: Refactor command queue buffer
Make the command struct a bit clearer by giving it an explicit
enum cmd_type instead of just a boolean to differentiate a
finish-fragment command from a release-gop command

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:05 +10:00
Jan Schmidt
bfdaae81f4 splitmuxsrc: Default to only keeping 100 files open
Add a reasonably large default for the number of simulataneous
files to open, that won't affect users that split recordings into
a few large files, but will help prevent fd exhaustion for users
that make recordings with lots of small fragments

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:04 +10:00
Jan Schmidt
1294264ab9 splitmuxsrc: Keep streams aligned during adjustments
When calculating the timestamp offset to apply to
media streams in a fragment, ensure that all fragments
are offset "together" to preserve alignment in cases
where there might gaps in a recording at a fragment boundary.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:04 +10:00
Jan Schmidt
682db96a41 splitmuxsrc: Add add-fragment signal and examples
Add a signal that allows adding fragments with a specific offset
and duration directly to splitmuxsrc's list. By providing the
fragment's offset on the playback timeline and duration directly,
splitmuxsrc doesn't need to measure the fragment making for faster
startup times.

Add a bus message that's published when fragments are measured,
reporting the offset and duration, so they can be cached by an
application and used on future invocations.

Add examples for handling the bus message and using the 'add-fragment'
signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:04 +10:00
Jan Schmidt
1821b52dd5 splitmuxsrc: Add num-open-fragments property
Add a property to limit the number of parts splitmux will open
simultaneously. Modify the part handling to support deactivating
and reactivating the demuxing for each part.

The default is '0', to preserve the existing behaviour of opening
all parts at the beginning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:04 +10:00
Jan Schmidt
eeb5a42b5d splitmuxsrc: Report minimum timestamp for each media stream
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7053>
2024-08-02 15:30:04 +10:00
Nicolas Dufresne
5df658cfdd qt6: glwindow: Don't leak previously rendered buffer
If the consumer reads the buffers too slowily, simply unref the
previously rendered buffer instead of leaking it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7286>
2024-08-01 12:44:06 +00:00
Shengqi Yu
7576d14762 v4l2object: append non colorimetry structure to probed caps
If the stream has a special colorimetry that is not in the colorimetry
list, it will cause negotiation to fail. We should allow passing any
colorimetry, so add an extra structure without the colorimetry field.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7029>
2024-07-31 09:28:18 +00:00
Hou Qi
5dffbd492c v4l2: Fix colorimetry mismatch for encoded format with RGB color-matrix
video-info supports encoded format to have RGB color-matrix, while
v4l2object just leave the v4l2 matrix to default when mapping
GST_VIDEO_COLOR_MATRIX_RGB. It causes gst matrix changed to be
GST_VIDEO_COLOR_MATRIX_BT601 when mapping v4l2 colorimetry.

So add support for encoded format with RGB color-matrix in v4l2object.
Note that for M2M encoders, we should in theory assume that that we can
transfer this value from OUTPUT to CAPTURE queues, though its only true
if the drivers does not do CSC. For now, we don't support any RGB
codecs, but leaving a note for the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3952>
2024-07-30 20:26:06 +00:00
Nicolas Dufresne
1ddb8797b5 v4l2object: SRGB colorspace is documented limited-range
Split JPEG and SRGB so that we can follow the specified difference. The
SRGB definition in V4L2 does not follow the standard, and is document
so. This is also why JPEG colorspace exists.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3952>
2024-07-30 20:26:06 +00:00
Nicolas Dufresne
20eb14b85b v4l2object: Fix size of plane_size array calculation
Due to missing parenthesys, only the first element of the array was
being cleared. As it is a staticly sized array in the object, this
code could also be simplified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3952>
2024-07-30 20:26:06 +00:00
Nicolas Dufresne
152df21644 v4l2object: Fix translation of quantization
The V4L2_MAP_QUANTIZATION macro has been fixed to something a lot saner,
fix our replica accordingly. The new macro now simply set the quantization
to full range is the pixel formats is RGB based, or if the JPEG
colorspace is used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3952>
2024-07-30 20:26:06 +00:00
Jan Schmidt
0faec707a0 adaptivedemux: Fail cleanly if parsebin is not installed
Detect a failure to construct a parsebin and error out
cleanly instead of trying to operate on a null pointer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6969>
2024-07-30 00:06:50 +00:00
Jan Schmidt
213726ca41 adaptivedemux2: Post a bus error when failing to start download
If a download completely fails to start, due to malformed URI or so,
post a bus error instead of just stalling out with no indication
why.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6969>
2024-07-30 00:06:50 +00:00
Jan Schmidt
f2a18ab277 adaptivedemux2: Implement file:// URI handling
Add the ability to play HLS and DASH from local files

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6969>
2024-07-30 00:06:50 +00:00
Jan Schmidt
ef0e822559 hlsng: Check caps are not null after parsing HLS CODECS tag
If the mime codec wasn't recognised, caps will be NULL and cause
a critical

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6969>
2024-07-30 00:06:50 +00:00
Loïc Yhuel
13034cc63f meson: fix SIZEOF_OFF_T when cross-compiling with Meson >= 1.3.0
https://mesonbuild.com/Release-notes-for-1-3-0.html#clarify-of-implicitlyincluded-headers-in-clike-compiler-checks

With only stddef.h, off_t is not defined, so when cross-compiling SIZEOF_OFF_T is -1.
We now use sys/types.h which should define off_t.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7217>
2024-07-23 15:32:22 +02:00
Sebastian Dröge
a786c85c4f taginject: Modify existing tag events of the selected scope
Not doing so would mean that tags would be overidden by any tag events sent by
upstream. Also only send a tag event directly if upstream never sent one.

By default use GST_TAG_MERGE_REPLACE to override tags that exist in both the
upstream event and this element with the ones from this element, but provide a
new "merge-mode" property to adjust the behaviour.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7145>
2024-07-10 13:00:34 +00:00
Sebastian Dröge
a36b3d9fcd taginject: Add getters for the properties
There's no reason why they should be write-only.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7145>
2024-07-10 13:00:34 +00:00
Sebastian Dröge
2ed84fe298 taginject: Use proper GType macro for the GstTagScope enum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7145>
2024-07-10 13:00:33 +00:00
Tihran Katolikian
cc1d978d7f qt6: explicitly specify path to QtGui private headers when including qrhi_p.h
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7131>
2024-07-04 09:52:57 +00:00
Matthew Waters
5bbeccbb65 qml/glsink: also support GLES2 needing shader 'precision' directives
Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3616
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7024>
2024-07-02 12:54:59 +00:00
Shengqi Yu
1875c178cd v4l2object: use v4l2 reported width for padded_width when complex video formats
Stride means bytes per line, and padded_width means pixels. Here,
padded_width shoule be pix width reported by v4l2 instead of stride.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7076>
2024-07-01 18:53:04 +00:00
Xavier Claessens
84b3a0950d build: Add missing common options that are yielding in subprojects
- Align `glib_debug`, `glib_assert` and `glib_checks` options with GLib,
  otherwise glib subproject won't inherit their value. Previous names
  and values are preserved using Meson's deprecation mechanism.
- Add `extra-checks` and `benchmarks` options in the main project so it
  can be inherited in GStreamer subprojects.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1165>
2024-06-27 15:53:46 +00:00
Jan Schmidt
73480e60d0 adaptivedemux: Fix handling closed caption streams
Fix a typo "CLOSED_CAPTION" -> "CLOSED-CAPTION" and
a broken if statement that always bailed out for
closed captions

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6985>
2024-06-26 13:44:27 +00:00
Tim-Philipp Müller
8d845d4a02 rtpdtmfsrc: minor logging clean-up
Only serialise event structure for debug logging purposes
if logging is actually enabled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7060>
2024-06-19 07:32:49 +00:00
Tim-Philipp Müller
62047a9f8d rtpdtmfsrc: fix leak when shutting down mid-event
.. and update rtpdtmfdepay unit test to trigger
the potential leak more reliably (without the fix).

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3633

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7060>
2024-06-19 07:32:49 +00:00
Tim-Philipp Müller
7d05af9680 rtpdtmfdepay: add unit test for caps fixation issue with downstream audioconvert
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7036>
2024-06-18 00:11:28 +01:00
Tim-Philipp Müller
ab61233f30 rtpdtmfdepay: fix caps negotiation with audioconvert
Specify "layout" field in src template to make sure it's
set and gets fixated properly if the downstream element
supports both interleaved and non-interleaved caps.

Fixes

  gst_pad_set_caps: assertion 'caps != NULL && gst_caps_is_fixed (caps)' failed

critical with e.g.

  gst-launch-1.0 rtpdtmfsrc ! rtpdtmfdepay ! audioconvert ! fakesink

Not that the layout really matters in our case since we always
output mono anyway, but non-interleaved requires adding AudioMeta,
so this is the easiest fix.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7036>
2024-06-18 00:11:28 +01:00
Mathieu Duponchelle
a20ef245a0 rtspsrc: fix invalid seqnum assertions
Upon fatal errors the loop function will first post an error message
then push out an EOS event.

An application may react immediately to the error message by setting the
state of the pipeline to NULL, meaning by the time we push out the EOS
event PAUSED_TO_READY may have reset the seek seqnum to -1.

While this is harmless, the assertion when setting an invalid seqnum
isn't tidy, fix this by simply not resetting to INVALID as it serves no
practical purpose and the next READY_TO_PAUSED will select a new seqnum
anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7032>
2024-06-14 11:28:06 +02:00
Jakub Vaněk
0b65f667af v4l2src: Interpret V4L2 report of sync loss as video signal loss
Certain V4L2 drivers can report that a video receiver is seeing
some signal, but that it is unable to synchronize to it. IOW: the driver
can sometimes report V4L2_IN_ST_NO_SYNC and not report V4L2_IN_ST_NO_SIGNAL.

In particular, I've seen the tc358743 (HDMI-to-CSI2 converter) driver
sometimes report this when deployed to a fleet of embedded Raspberry Pis.
The relevant kernel code is in [1]. The video output is not practically
usable when V4L2_IN_ST_NO_SYNC is reported (only visually corrupted frames,
sometimes with random "snow", are received). I assume that this happens when
either the HDMI cable is poorly plugged in or damaged or when a CSI2 FFC
cable is used and is damaged.

The change in this commit is useful for detecting this working-but-not-really
condition in application code. Applications already listening for the "Signal lost"
message will gain the ability to handle this condition.

There seem to be more V4L2 error flags like this, see [2]. However, I do not
have practical experience with them and adding only V4L2_IN_ST_NO_SYNC seems
like a safer option.

[1]: https://github.com/raspberrypi/linux/blob/be8498ee21aa/drivers/media/i2c/tc358743.c#L1534
[2]: https://www.kernel.org/doc/html/v6.6/userspace-api/media/v4l/vidioc-enuminput.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7021>
2024-06-12 17:26:48 +00:00
Edward Hervey
98e4d90519 adaptivedemux2: Don't send FLUSH_{START|STOP} when losing sync
The initial goal was to support the case where we are paused watching a live
stream, and when we resume we can no longer resume from the previously
downloaded position. In that case we internally do a flushing seek back to the
"current live head position". This was also extended since to be able to
handle (utterly broken) servers when we can't really figure out where we are
anymore and therefore trigger that lost sync so we can try to get back on our
feet.

This does fix the issue... but results in spurious FLUSH_{START|STOP} events
being sent downstream. While that's fine for regular playback scenarios, it's a
bit of a wild scenario since a lot of pipelines/applications don't expect such
events when it wasn't triggered by downstream/application.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7005>
2024-06-12 06:05:24 +00:00
Sebastian Dröge
441e71d1ff flvmux: Use GDateTime instead of gmtime()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6872>
2024-06-06 08:33:51 +00:00
Corentin Damman
bdeabcc4a6 gstqsg6material: fix RGB format support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6991>
2024-06-05 16:49:06 +00:00
Chun-wei Fan
d024ee4303 GTK plugin: Support OpenGL/WGL on Windows
This attempts to implement the gtkglsink element on Windows using WGL,
as there were some more gotchas that are along the way, since we need to
juggle with libepoxy along the way, meaning that we need a recent
GTK+-3.24.x for this to work properly, i.e. the upcoming GTK+-3.24.43.

Since we are essentially using an overlay compositor only during
rendering, move its initialization and destruction into the
gtk_gst_gl_widget_render() function, so that things are safer as we are
doing things across threads between gstreamer (gst-gl) and GTK, as GL
operations, as above, have more gotchas on Windows.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4289>
2024-06-05 08:53:19 +00:00
Piotr Brzeziński
9ca8f16a3b macos: Listen for audio devices being added/removed
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6981>
2024-06-01 13:21:59 +00:00
Sebastian Dröge
9b60b32cf8 rtspsrc: Only update from the Content-Base header in the initial OPTION / DESCRIBE response
Some servers send a new content base in the SETUP response, which is
just the non-aggregate control URL of the individual streams.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:44 +00:00
Sebastian Dröge
e65344afac rtspsrc: Handle the case of * as session-wide control URL from the SDP
Just like the comment above says this is supposed to indicate that the
same URL should be used as for the connection so far. If encountering
this case simply do nothing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:44 +00:00
Sebastian Dröge
e73e34fd6f rtspsrc: Also handle rtsps:// and similar URLs as absolute in other places
Previously a direct comparison with `rtsp://` was performed, which
didn't catch cases like `rtsps://`.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:44 +00:00
Sebastian Dröge
966c39b92e rtspsrc: Don't try the SETUP workaround for broken servers with absolute control URIs
Previously only control URIs that started with "rtsp://" were ignored
but it makes more sense to ignore all absolute URIs.

gst_uri_is_valid() conveniently checks for exactly that.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3563

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6926>
2024-06-01 11:30:43 +00:00
Seungha Yang
fd21d97060 qtdemux: Handle keyunit trick mode in case of push mode too
Skip non-keyframe video frames if trickmode-keyunit flag is set

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5900>
2024-05-31 11:21:55 +00:00
Seungha Yang
05f9eadcaf qtmux: Handle time information value > UINT32_MAX
If any duration in timescale is larger than UINT32_MAX, use version 1
atom, otherwise file header will be constructed with truncated values.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6843>
2024-05-28 16:09:58 +00:00
Edward Hervey
c924e4cc1e hlsdemux2: Minor refactoring of starting segment check
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
5bc9883d68 hlsdemux2: Be more tolerant when matching segments with PDT
Some servers might not provide 100% matching PDT when doing updates, or accross
variants. This would cause the code matching segments using PDT to fail if the
segment PDT was 1 microsecond (or whatever small value) before the candidate
segment. And would pick the (wrong) following segment as the matching one.

In order to be more tolerant when matching, we instead check whether the
candidate segment is within the first segment of the segment we are trying to
match.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
81fd460c90 hlsdemux2: Fix failure to find a replacement segment on resync
If we end up with a segment with an internal time that varies from the supposed
one, this could be for two reasons:
* We guess-timated the wrong segment to go to when advancing or switching
  variants. In that case we try to find the actual segment to go to (just before
  this change).
* There was a complete playlist change (for whatever reason) and we can't find a
  replacement. In that case we want to carry on playback from this position but
  need to remember that we moved (by setting the stream to DISCONT, and
  resetting the new mapping).

Fixes playback on several broken stream

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
3e810a6721 hlsdemux2: Refactor update of GstHLSTimeMap values
This was also missing transferring the PDT if present

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
9a7f455aea hlsdemux2: Fix parsing of EXT-X-DISCONTINUITY-SEQUENCE:0
Since the default value of `m3u8->discont_sequence` (before parsing of the
playlist data) was 0 .. we would never properly detect the presence of that
field if it was present with a value of 0.

This would later on cause havoc in playlist synchronization where we would
assume it didn't have a discontinuity sequence specified (whereas it did, and it
was 0).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
d2b3262b71 hlsdemux2: Increase tolerance for discontinuity detection
A lot of streams will do a poor job of estimating proper duration of fragments
in the playlist, but over several fragments have it correct.

Instead of constantly trying to realign the estimated stream time, allow for a
more realistic tolerance of 3-4 video frames

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
8b6e7a018c hlsdemux2: Ensure a discont will be set when resetting for lost sync
This is to ensures we inform the demuxer/parsers that what follows is not contiguous

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
836bca461a hlsdemux2: Fix handling of variant switching and playlist updates
When updating playlists, we want to know whether the updated playlist is
continuous with the previous one. That is : if we advance, will the next
fragment need to have the DISCONT buffer set on it or not.

If that happens (because we switched variants, or the playlist all of a sudden
changed) we remember that there is a pending discont for the next fragment. That
will be used and resetted the next time we get the fragment information.

Previously this was only partially done. And it was racy because it was set
directly on `GstAdaptiveDemux2Stream->discont` when a playlist was updated,
instead of when the next fragment was prepared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
7d49b1cc51 adaptivedemux2: Only set DISCONT on beginning of fragments
This avoids accidentally setting it in the middle of a fragment, which could
cause havoc in demuxer/parsers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Edward Hervey
81c42ee14b hlsdemux2: Fix getting starting segment on live playlists
When dealing with live streams, the function was assuming that all segments of
the playlist had valid stream_time. But that isn't TRUE, for example in the case
of failing to synchronize playlists.

Fixes losing sync due to not being able to match playlist on updates

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6610>
2024-05-28 14:00:57 +00:00
Sebastian Dröge
9156b373e6 rtpbin: Regularly emit the sync signal
Even if no new synchronization information is available.

This is necessary because the timestamp offset logic in rtpbin depends
on the base RTP time that is determined by the jitterbuffer, but this
changes all the time (especially in mode=slave) and the timestamp
offsets have to be updated accordingly. Doing so is especially important
if they're only determined by the RTP-Info, which never changes from the
very beginning.

The interval can be configured via the new min-sync-interval property.
Synchronization happens at least that often, but at most as often as the
old sync-interval property allows.
Both intervals are now based on the monotonic system clock.

Additionally, clean up synchronization code a bit, only emit either
inband NTP or RTCP SR synchronization at the same time, based on which
one has the more recent time information, and only emit RTP-Info
synchronization if it wasn't provided previously at the same time as the
NTP-based synchronization information.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:31 +00:00
Sebastian Dröge
df8c29e340 rtpjitterbuffer: Set max-rtcp-rtp-sync-time to -1 (disabled)
There is generally no requirement to ignore RTCP SR if the RTP time of
the SR differs a lot from the last received RTP packet. The mapping
between RTP and NTP time stays valid until there was a stream reset, in
which case we wouldn't use that information anyway.

When using rtcp-sync-send-time=false the default of 1s difference can
easily be exceeded, e.g. if encoding of the stream after capture adds
more than 1s of latency.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
95a0649945 rtpbin: Allow synchronizing against RTP-Info without having received any RTCP
Previously the information was provided from rtpjitterbuffer to rtpbin
only once the first RTCP SR was received, which is not necessary at all
as all required information is available from the caps already.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1162

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
8bfba72ea4 rtpbin: Add new never/ntp RTCP sync modes
Never is useful for some RTSP servers that report plain garbage both via
RTCP SR and RTP-Info, for example.

NTP is useful if synchronization should only ever happen based on RTCP
SR or NTP-64 RTP header extension.

Also slightly change the behaviour of always/initial to take RTP-Info
based synchronization into account too. It's supposed to give the same
values as the RTCP SR and is available earlier, so will generally cause
fewer synchronization glitches if it's made use of.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
158f12b5da rtpbin: Handle switches between RTP-Info and NTP-based stream association better
Instead of switching on the very first stream, require that all streams
have switched before switching to the different synchronization
mechanism.

Without this there will be a noticeable gap during the switch. E.g. when
going from RTP-Info to NTP-based association, first the first stream
only would get an offset, then the first two, ... then all of them.
Depending on the order of streams this will cause a lot of changes in
ts-offset during the transition.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
b30671a8ee rtpbin: Pass NPT start from rtpjitterbuffer to rtpbin
And use it to detect synchronization changes (e.g. seeks) more reliably
when doing RTP-Info based synchronization.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
3eb22af88b rtpbin: Clean up stream association state
Use fewer magic numbers and keep track of the different synchronization
mechanisms separately. Also keep track of more state to detect more
situations when resynchronization should happen.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
d8dabf142f rtpbin: Constify function parameters and use correct types
Previously these parameters were randomly changed in the body of the
function to avoid having to declare a new variable, which made the code
very hard to follow. By marking them as const this won't be possible
anymore in the future.

Also the RTP clock-base (RTP time from RTSP RTP-Info) is an unsigned
64 bit integer as it's an extended RTP timestamp.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
155c3fb3b2 rtpbin: Untangle NTP-based and RTP-Info based stream association
Both were entangled previously and very hard to follow what happens
under which conditions. Now as a very first step the code decides which
of the two cases it is going to apply, and then proceeds accordingly.
This also avoids calculating completely invalid values along the way and
even printing them int the debug output.

Also improve debug output in various places.

This shouldn't cause any behaviour changes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
7d0c7144ba rtpbin: Remove unused variable / function parameter
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
4421c3de75 rtpbin: Handle ntp-sync=true before everything else
This simplifies the code as it's a much simpler case than the normal
inter-stream synchronization, and interleaving it with that only
reduces readability of the code.

Also improve some debug output in this code path.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
4b0e75a094 rtpbin: Add some documentation to gst_rtp_bin_associate()
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sebastian Dröge
70a435c0c4 rtpbin: Don't do any timestamp offsetting in rfc7273-sync=true mode
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1160

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6543>
2024-05-28 11:52:30 +00:00
Sergey Krivohatskiy
1c5e1798b6 flacparse: fix buffer overflow in gst_flac_parse_frame_is_valid
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6835>
2024-05-27 23:31:44 +00:00
Tim-Philipp Müller
8bd1a3213e level: fix old "message" property doc chunk
In the online documentation the new post-messages
property would show up as deprecated refering to
itself.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6911>
2024-05-23 21:36:37 +00:00
Sebastian Dröge
cd606696a6 gtk: Fail initialization of the sink if GTK4 is already initialized in the same process
Initializing GTK3 and GTK4 in the same process does not work and is not
supported.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6892>
2024-05-23 08:15:44 +00:00
Piotr Brzeziński
477beab403 osxaudio: Avoid using private APIs on iOS
Turns out AudioConvertHostTimeToNanos and AudioGetCurrentHostTime are macOS-only APIs, which prevents apps using
GStreamer on iOS from being accepted into App Store.

This commit replaces those functions with a manual version of what they do - mach_absolute_time() for the current time,
and data from mach_timebase_info() at the beginning to convert host timestamps to nanoseconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6789>
2024-05-22 08:58:24 +00:00
Diego Nieto
453a6f1800 rtsp-server: Remove unused define in backchannel test
The caps match with the ones used in test-onvif-backchannel,
but they are actually not used here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6885>
2024-05-21 13:25:44 +02:00
Jan Schmidt
64133b40a7 rtpmp4gdepay: Set duration on outgoing buffers
If we have constant duration buffers, set the duration on
outgoing buffers, like rtpmp4adepay does. This fixes
problems with (for example) muxers like mp4mux not writing
the duration of the final sample into the index.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6878>
2024-05-20 15:24:32 +00:00
Guillaume Desmottes
210487b50a wavparse: reset when receiving STREAM_START
We need to reset the internal state to be able to parse a new stream.
When doing so keep seek event and do not destroy the adapter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6840>
2024-05-16 11:35:02 +00:00
Sebastian Dröge
8ea355e52c audioringbuffer: Avoid overflows of segment done counter
This counter is incremented once for every segment, meaning it would
e.g. overflow after 24 days when using 1ms segments. Once that happens,
completely wrong positions are reported and invalid memory is handed out
for writing/reading the next segments.

As the affected variables are unfortunately part of the public API of
the struct, a second set of variables is added together with accessor
functions and both variables are kept in sync for backwards
compatibility.

All existing users of the two variables are moved to the new ones but
external code might still run into the overflow.

This also slightly breaks API as external code updating the variables
will have no effect anymore but the only known user of this is
pulsesink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6740>
2024-05-16 06:52:58 +00:00
Sebastian Dröge
a4514c5458 level: Don't post a message on EOS without a valid audio info
If EOS is received before caps, e.g. because of an error, then rate and
number of channels would be 0 and some divisions by zero would happen.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6819>
2024-05-12 07:06:32 +00:00
Sebastian Dröge
0ef396359c gst: Move GstQueueArray as GstVecDeque to core
And change lengths and indices from guint to gsize for a more correct type.

Also deprecate GstQueueArray and implement it in terms of GstVecDeque.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6779>
2024-05-06 18:25:42 +00:00
Sebastian Dröge
efba52fcba qtdemux: Use G_GUINT64_CONSTANT when creating test caps
Otherwise this fails on 32 bit platforms.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3521

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6804>
2024-05-06 06:18:35 +00:00
Seungha Yang
c8d01d7d1a video: Add Y216 and Y416 formats
The same memory layout as Y212 and Y412 formats, respectively

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6745>
2024-05-03 17:02:34 +00:00
Tim-Philipp Müller
eec64e372b rtph264depay: fix FU-B handling
Skip extra 16-bit DON in FU-B header.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/806

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6607>
2024-04-29 12:21:52 +00:00
Tim-Philipp Müller
b1a45b527a rtph264depay: minor refactoring of FU handling code
Make code easier to follow, and prepare for next commit.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6607>
2024-04-29 12:21:52 +00:00
William Wedler
9ad6a9b942 fix: qmlglsink: video content resizes to new item size
Mark geometry dirty when the item rectangle changes in the
QtGLVideoItem::updatePaintNode method. This allows changes in the bounding
rectangle to be applied to the scene graph geometry node.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6711>
2024-04-29 02:57:06 +00:00
William Wedler
c02af39026 fix: qml6glsink: video content resizes to new item size
Mark geometry dirty when the item rectangle changes in the
QtGLVideoItem::updatePaintNode method. This allows changes in the bounding
rectangle to be applied to the scene graph geometry node.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6711>
2024-04-29 02:57:06 +00:00
Tim Blechmann
ff7b41ac86 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/6736>
2024-04-26 09:45:49 +08:00
Edward Hervey
4e5a54612e adaptivedemux2: Answer GST_QUERY_CAPS
If we have a generic caps, we can answer the query.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6690>
2024-04-23 07:09:21 +00:00
Edward Hervey
6b43e4e19f adaptivedemux2: Refactor output slot creation
Set as much information as possible on the slot (including the associated
track) *before* the associated source pad is added to the element.

We need this so that incoming event/queries can be replied to if they are
received when adding the pad

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6690>
2024-04-23 07:09:21 +00:00
Philipp Zabel
46a41667a3 v4l2bufferpool: Ensure freshly created buffers are not marked as queued
Otherwise, if we run in to the copy case, this can cause these
groups to stay around with queued flag set, but never actually
queued, until gst_v4l2_allocator_flush() is called, which then
erroneously frees the associated memories, causing the release
function to decrement the allocator refcount where it was never
incremented, resulting in early allocator disposal, and either
deadlock or use after free.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6552>
2024-04-18 16:42:43 +00:00
Qian Hu (胡骞)
8d003f00e9 v4l2: add multiplane y42b(yuv422m)
for some jpg file, mediatek v4l2 jpeg decoder
hardware produce multi plane YUV 4:2:2 data

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6617>
2024-04-16 09:03:47 +00:00
Hou Qi
105d232fde v4l2bufferpool: queue back the buffer flagged LAST but empty
Some decoder drivers need to wait enough capture buffers before
starting to decode. But the dequeued buffer flag LAST but empty
has no chance to queue back to driver, which makes decode hang
after seek. So need to queue back such kind of buffer to driver.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6579>
2024-04-15 18:07:17 +00:00
Philipp Zabel
e1f5bacf8d v4l2: bufferpool: Drop writable check on output pool process
Output buffers don't have to be writable. Accepting read-only buffers
from the V4L2 buffer pool allows upstream elements to write directly
into the V4L2 buffers without triggering a CPU copy into a new buffer
from the same V4L2 buffer pool every time.

Tested with the vivid output device:

  GST_DEBUG=GST_PERFORMANCE:7 gst-launch-1.0 videotestsrc ! v4l2sink device=/dev/video5

With this change, gst_v4l2_buffer_pool_dqbuf() must be allowed to not
resize read-only memories of output buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6572>
2024-04-15 17:11:00 +00:00
Philippe Normand
111cc8d796 vpxdec: Include vpx error details in errors and warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6626>
2024-04-13 10:57:43 +01:00
Philippe Normand
bd83046193 vp9enc: Include vpx error details in errors and warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6626>
2024-04-13 10:56:29 +01:00
Philippe Normand
73ce4fd770 vpxenc: Rename GST_VPX_WARN to GST_VPX_ENC_WARN
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6626>
2024-04-13 10:55:55 +01:00
Qian Hu (胡骞)
cd95d02032 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/6616>
2024-04-12 16:59:33 +08:00
William Wedler
942415dce0 fix: qml6glsink: Notify that the returned QSGNode node has changes
Sets the QSGNode::DirtyMaterial bit when a new buffer is used for the material's texture

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3469
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6609>
2024-04-11 14:21:04 +00:00
Jochen Henneberg
687b3a2027 qt6: Let plugin documentation show up
* Added qml6 to plugin cache
* Added 'since' markers
* Moved plugin to plugins-good where it really is
* Fixed section comments

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6603>
2024-04-11 12:38:59 +00:00
Jochen Henneberg
8b87d7bcf7 qt: Let plugin documentation show up
* Enabled cc file parsing from hotdoc
* Moved package to plugins-good where it really is
* Fixed section comments

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6603>
2024-04-11 12:38:59 +00:00
Jochen Henneberg
fee46dee28 qt6: Added support for NV12 input format to qml6glsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6582>
2024-04-10 13:45:26 +02:00
Jochen Henneberg
7065d540ee qt: Added support for NV12 input format to qmlglsink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6582>
2024-04-10 13:45:26 +02:00
Jimmy Ohn
a6c8c6f866 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/6558>
2024-04-10 08:43:56 +00:00
Arun Raghavan
82b10e57b0 pulsesink: Re-enable emission of stream status messages
This was disabled almost 10 years ago because we were missing libpulse API to
avoid a deadlock. That was fixed quite a long time ago, so let's enable this
again. The defer counter becomes an atomic, as we no longer have a threaded
mainloop lock protecting it.

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3444
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6500>
2024-04-09 15:50:04 +00:00
Philippe Normand
8d3e7689e1 vpxenc: Include vpx error details in errors and warnings
The vpx_codec_t err_detail string usually provides additional context about the
error, so include it in GStreamer warnings and errors, when it's not NULL.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6573>
2024-04-09 14:40:21 +00:00
Jochen Henneberg
6e33a5da14 qt6: Fixes for dummy texture
* RED_OR_ALPHA8 will map value to alpha for OpenGL, use R8 to avoid
  2nd shader
* Determine texel size for proper texture memory preparation
* QByteArray::fromRawData() does shallow copy and thus leads to use of
  corrupted memory
* Make sure RGBA dummy texture is fully opaque
* QRhiTexture::create() must be called to allocate texture resources

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6578>
2024-04-08 20:05:10 +02:00
Jochen Henneberg
87dc22b053 qt: Fixup for dummy textures
* Initialize dummy texture Ids
* Ensure YUV->RGB matrix set for dummy textures

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6578>
2024-04-08 20:05:09 +02:00
Sebastian Dröge
0596871b98 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/6536>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
11ce209ea0 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/6536>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
0c34c85f7a 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/6536>
2024-04-08 10:29:54 +00:00
Sebastian Dröge
4a4eb56fc2 rtspsrc: Optionally timestamp RTP packets with their receive times in TCP/HTTP mode
Until https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509
this was accidentally done inside rtpjitterbuffer for many years, and
doing so potentially solves problems on some streams while introducing
problems on others.

Make this configurable on rtspsrc and default to not set timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6529>
2024-04-08 08:34:38 +00:00
Jan Schmidt
832a517965 rtpjitterbuffer: Don't use estimated_dts to do default skew adjustment
When the buffer DTS is estimated based on arrival time at the
jitterbuffer (rather than provided on the incoming buffer itself),
it shouldn't be used for skew adjustment. The typical case is
packets being deinterleaved from a tunnelled TCP/HTTP RTSP stream,
and the arrival times at the jitter buffer are not well enough
correlated to usefully do skew adjustments.

This restores the original intended behaviour for the 'estimated dts'
path, that was broken years ago during other jitterbuffer refactoring.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6509>
2024-04-07 12:24:58 +00:00
Sebastian Dröge
ee566b8960 flac: Add wrap file and add fallback for it to the flac plugin
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6553>
2024-04-07 11:12:51 +00:00
Tim Blechmann
1c9fe19b23 v4l2: enforce a pixel aspect ratio of 1/1 if no data are available
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6242>
2024-04-07 10:14:18 +00:00
Philipp Zabel
6f9872cb56 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/6551>
2024-04-06 11:44:27 +00:00
Elliot Chen
e4ee4ca716 v4l2: fix error in calculating padding bottom for tile format
This is a regression while porting to arbitrary tile dimensions
introduced in .

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6480>
2024-04-05 13:28:47 +00:00
Elizabeth Figura
c308f013a7 atdec: Handle channel counts greater than 2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6157>
2024-04-05 06:54:24 +00:00
Elizabeth Figura
277d6ddf22 atdec: Use gst_audio_decoder_set_output_caps() directly
The code currently sets the same caps in two different ways, and neither of them correctly handle the channel mask.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6157>
2024-04-05 06:54:24 +00:00
Sebastian Dröge
16f69acf30 wavpackparse: Use an unsigned integer for the block size calculations
It's never negative.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Sebastian Dröge
eefb7c1638 wavpackparse: Fix potential integer overflow on ID_ODD_SIZE blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Sebastian Dröge
6402978a48 wavpackparse: Explicitly handle ID_WVX_NEW_BITSTREAM
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6498>
2024-04-04 15:10:02 +00:00
Robert Guziolowski
52638c1b22 qml6glsink: fix destruction of underlying texture
One should not directly delete the QRhiTexture instance.
Instead it should be marked as to be deleted once QRhi::endFrame()
is called (see: https://doc.qt.io/qt-6/qrhiresource.html#deleteLater )

Fixes: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3443
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6467>
2024-04-02 11:55:16 +11:00
Tim-Philipp Müller
ef5b8dc96a tests: rtpred: fix out-of-bound writes
Don't write more data to the buffer than we allocated
space for.

Fixes 

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6474>
2024-03-28 19:51:47 +00:00
Haihua Hu
37e3a38ba9 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/6451>
2024-03-28 12:53:01 +00:00
Jan Schmidt
351936aeac 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/6447>
2024-03-27 10:53:38 +00:00
Sebastian Dröge
e0dfb3d974 rtphdrext-ntp: Fix typo of the RFC number in the element metadata
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/3417

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6439>
2024-03-26 14:37:47 +02:00
Hou Qi
024d3ab051 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/6416>
2024-03-22 16:02:51 +00:00
Edward Hervey
5280f0b733 adaptivedemux2: Add libsoup tracing debug
Provides more information for debugging

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6409>
2024-03-20 09:48:12 +00:00
Edward Hervey
3d500636a9 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/6392>
2024-03-19 13:25:41 +00:00
Edward Hervey
ab11c20d59 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/6392>
2024-03-19 13:25:41 +00:00
Edward Hervey
46bb0bfa57 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/6392>
2024-03-19 13:25:41 +00:00
Edward Hervey
8438c3f567 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/6392>
2024-03-19 13:25:41 +00:00
Alexander Slobodeniuk
650534c940 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/6355>
2024-03-13 19:32:46 +00:00
Sebastian Dröge
38011a01dc 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/6318>
2024-03-13 12:48:36 +00:00
Piotr Brzeziński
e9802f5f41 macos: Add Apple AAC encoder (atenc)
Adds the `atenc` element capable of encoding AAC-LC audio, using the AudioToolbox framework.
It's able to encode up to 7.1 channel configurations.
Comes with basic knobs for rate control (bitrate for CBR, quality for VBR).

Support for more profiles (LD, HE-AAC) should be simple, but is not included here because of bugs
with parsing of the AudioSpecificConfig.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6254>
2024-03-12 19:50:06 +00:00
Piotr Brzeziński
d3fba31da0 macos: Move atdec from applemedia (-bad) to osxaudio (-good)
osxaudio has a few helper methods potentially useful in atdec (or future atenc), like GStreamer -> CoreAudio
channel mapping. Doesn't make sense to duplicate them in applemedia, and atdec is the only audio-oriented
element there anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6223>
2024-03-12 09:55:10 +00:00
Piotr Brzeziński
9c084faa75 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/6320>
2024-03-11 18:18:01 +00:00
Nirbheek Chauhan
3bed35c342 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/6302>
2024-03-11 09:15:50 +00:00
François Laignel
7d5bb1ea7a webrtc: add all SSRC attributes getting CAPS for a PT
The transport stream only returned the CAPS for the first matching PT entry
from the `ptmap`. Other SSRC with the same PT where not included. For a stream
which bundled multiple audio streams for instance, only the first SSRC was
knowed to the SSRC demux and downstream elements.

This commit adds all the `ssrc-` attributes from the matching PT entries.

The RTP jitter buffer can now find the CNAME corresponding its SSRC even if it
was not the first to be registered for a particular PT.

The RTP PT demux removes `ssrc-*` attributes cooresponding to other SSRCs
before pushing SSRC specific CAPS to downstream elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6119>
2024-03-08 10:28:15 +00:00
Michael Tretter
5b3082257e 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/6292>
2024-03-08 02:14:11 +00:00
Mathieu Duponchelle
519546aea3 rtpgstpay: flush on EOS
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5173>
2024-03-07 14:02:33 +00:00
Sebastian Dröge
b88d69b722 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/5173>
2024-03-07 14:02:33 +00:00
Elizabeth Figura
e2167867d5 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/6225>
2024-03-07 12:52:30 +02:00
Jan Schmidt
f53dbb28b2 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/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
57013e1a7c 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/6248>
2024-03-06 17:50:53 +00:00
Jan Schmidt
4d2f000125 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/6248>
2024-03-06 17:50:53 +00:00
Tim-Philipp Müller
4db25f1500 rtspsrc: Consider 503 Service Not Available when handling broken control urls
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6213>
2024-03-05 17:45:18 +00:00
Tim-Philipp Müller
756064b9c3 Back to development
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6261>
2024-03-05 12:58:57 +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