Commit graph

3834 commits

Author SHA1 Message Date
Mengkejiergeli Ba
9147dc3d37 vaapidec: Fix uninitialized vars
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3918>
2023-02-16 03:47:12 +00:00
Thibault Saunier
2837f14b6b tests: Move srtp test known issue def to the 'check' testuite
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3979>
2023-02-16 02:39:05 +00:00
Thibault Saunier
1146548c3d validate:launcher: try to bne more explicit about where to add known failure
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3979>
2023-02-16 02:39:04 +00:00
Thibault Saunier
c637d90be9 validate:launcher: Handle known return code issue in none validate based tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3979>
2023-02-16 02:39:04 +00:00
Thibault Saunier
47a07a4f5a uri-asset: Don't create AudioSource asset for unknown track types
And just skip them as we can't do anything else with them right now

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3976>
2023-02-16 01:47:28 +00:00
Luke McGartland
396c758544 Update subprojects/gst-editing-services/ges/ges-uri-asset.c
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3976>
2023-02-16 01:47:28 +00:00
Hosang Lee
0efb792fb4 tests: qtdemux: add test for MSS fragment wrong data offset compensation
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams. The samples will not be located and
eventually playback will error out. So compensate assuming data
is in mdat following moof.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Tim-Philipp Müller
491feead6e tests: qtdemux: use binary files for samples
Instead of hexdumping it in a 360k header file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Hosang Lee
88f16ebd2a qtdemux: compensate wrong data offset for MSS fragments
A data offset with an offset smaller than the moof length is wrong
in smooth streaming streams.

The samples will not be located and eventually playback will
error out. So compensate assuming data is in mdat following moof.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3840>
2023-02-16 00:43:57 +00:00
Seungha Yang
f7c2602d41 splitmuxsrc: Proxy latency query to part reader
splitmuxsrc can respond to the latency query

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3566>
2023-02-15 23:47:50 +00:00
Ruben Gonzalez
75b75826dc validate:ssim: fix gst-validate tool name in the example
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3974>
2023-02-15 23:06:50 +00:00
Mathieu Duponchelle
e728c0e71a ges-source: make sure internal decodebin doesn't get seeked too early
When uridecodebin exposes pads for its streams, we immediately ghost
the relevant (selected) one and let composition send a seek as soon as a
buffer is probed.

This means that sometimes uridecodebin is still linking elements
internally (for non-selected streams) and sees flush events travel down
the elements it is still busy trying to link / forward sticky events to.

This causes all sorts of nasty issues, which can be avoided by simply
blocking all data flow from the source until no-more-pads has been
emitted by uridecodebin (or whatever sub_element is wrapped).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3971>
2023-02-15 22:24:51 +00:00
Mathieu Duponchelle
2fd2af9420 ges-source: do not emit no-more-pads on child element
This most likely never caused any issues as we don't connect to
no-more-pads in the first place, and the element isn't directly exposed
to the user, but emitting it makes no sense, and we are actually going
to connect to no-more-pads in a subsequent commit.

The call was added in 86b893e54c, a patch
by me in 2013, I have no idea why but I probably didn't have a firm
grasp on what I was doing then.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3971>
2023-02-15 22:24:51 +00:00
Mengkejiergeli Ba
585d2d93f6 codecparsers: {h264,h265}bitwriter: Remove redundant condition checks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3967>
2023-02-15 16:00:04 +08:00
Khem Raj
817339c4de v4l2: Define ioctl_req_t for posix/linux case
this is an issue seen with musl based linux distros e.g. alpine [1]
musl is not going to change this since it breaks ABI/API interfaces
Newer compilers are stringent ( e.g. clang16 ) which can now detect
signature mismatches in function pointers too, existing code warned but
did not error with older clang

Fixes
gstv4l2object.c:544:23: error: incompatible function pointer types assigning to 'gint (*)(gint, ioctl_req_t, ...)' (aka 'int (*)(int, unsigned long, ...)') from 'int (int, int, ...)' [-Wincompatible-function-pointer-types]
    v4l2object->ioctl = ioctl;
                      ^ ~~~~~

[1] https://gitlab.alpinelinux.org/alpine/aports/-/issues/7580

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3950>
2023-02-14 20:36:28 +00:00
Fernando Jiménez Moreno
1cabfebd2f docs: fix gst-validate-launch test suite example.
According to gst-validate-1.0 --list-scenarios, play_5s is not
a valid scenario. Runnning the test suite example with it
ends up raising an AttributeError. Switching to play_15s makes
it work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3941>
2023-02-14 19:49:04 +00:00
Evgeny Pavlov
351b72a59a amfcodec: Initial support of Smart Access Video
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3904>
2023-02-14 19:04:30 +00:00
Evgeny Pavlov
38c3de565b amfcodec: Update AMF headers to version 1.4.29
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3904>
2023-02-14 19:04:30 +00:00
Tim-Philipp Müller
7cb9cea2b5 validate: add elements_srtp.test_play_key_error to flaky test list
See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1777

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3959>
2023-02-14 18:27:12 +00:00
U. Artie Eoff
233de05a98 avviddec: change AV_CODEC_CAP_AUTO_THREADS->AV_CODEC_CAP_OTHER_THREADS
This fixes a compile error with recent upstream FFmpeg.

The AV_CODEC_CAP_AUTO_THREADS was deprecated and renamed to
AV_CODEC_CAP_OTHER_THREADS in FFmpeg upstream commit
7d09579190de (lavc 58.132.100).

The AV_CODEC_CAP_AUTO_THREADS was finally removed in FFmpeg upstream
commit 10c9a0874cb3 (lavc 59.63.100).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3951>
2023-02-14 09:01:04 -05:00
Sebastian Dröge
cfb14c4b5d svtav1enc: Drain and completely reallocate the encoder instance on caps changes
The encoder does not support reconfiguration, and only deinitializing it
and then initializing it again causes deadlocks.

Also only reconfigure and drain the encoder if the video info has
actually changed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3957>
2023-02-14 11:59:48 +00:00
Yang, Xuchen
a14f8008fb audio: channel-mix: Fix channel count limit to be able to equal 64
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3949>
2023-02-14 07:04:45 +00:00
Thibault Saunier
dc1c9c8985 glvideomixer: Keep a reference to the underlying pad
There was cases where we were accessing the pad while it was already
destroyed. Ensure it can't happen by owning a ref on it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3916>
2023-02-13 22:23:13 +00:00
Piotr Brzeziński
04615ca55a vtenc/vtdec: Fix typo in RGBA64_LE availability define
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3912>
2023-02-13 17:48:37 +00:00
Piotr Brzeziński
8d81c4827c vtenc: Disable ARGB/RGBA64 caps on M1 Pro/Max with macOS <13
Fixes #1358.
Passing ARGB64/RGBA64 to vtenc caused the encoding to fail
when running on M1 Pro/Max variants with macOS 12.x, so let's
remove these formats from caps when such scenario is detected.
This issue appears to have been fixed OS-side in macOS 13.0.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3912>
2023-02-13 17:48:37 +00:00
Evgeny Pavlov
cf9a82cf84 amfcodec: Add 10-bit & HDR support in AMF AV1 encoder
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3641>
2023-02-13 15:38:07 +00:00
Evgeny Pavlov
ec7eb75312 amfcodec: Add 10-bit & HDR support in h265 encoder
This commit adds 10 bit source file encoding and HDR metadata support in AMF HEVC encoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3641>
2023-02-13 15:38:07 +00:00
Edward Hervey
15bb6abeca gst-play: Don't force accurate seeking
This crept in several years ago sadly :(

The usage of accurate seeking should be reserved to use-cases where it is
essential that we seek to that position. This should not be the default.

There is a new option `--acurate-seeks/-a` to be able to force that.

Furthermore, if accurate seeks aren't required, a player should be using the
GST_SEEK_FLAG_KEY_UNIT flag to seek to the closest keyframe and provide the most
reactive experience.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3926>
2023-02-13 09:25:31 +00:00
Seungha Yang
b3acafc2b0 amfav1enc: Set stream-format on caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3946>
2023-02-12 23:18:25 +09:00
Seungha Yang
13ae9d2067 qsvav1enc: Set stream-format on caps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3946>
2023-02-12 23:18:04 +09:00
Jan Vermaete
c99db9a42b rtsp-server: test-launch: added the argument mount/m to specify the mountpoint
Default is the previous hardcoded "/test"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3942>
2023-02-12 09:21:19 +01:00
Vivia Nikolaidou
4e7a5ebb11 qtdemux: Handle moov atom length=0 case by reading until the end
Previously it would fail to demux the file by trying to read G_MAXUINT64
bytes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3934>
2023-02-11 02:20:39 +00:00
Vivia Nikolaidou
3a9acff978 qtdemux: Fix guint vs gsize type confusion
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3934>
2023-02-11 02:20:39 +00:00
Piotr Brzeziński
e2deebb103 vtdec: Fix not waiting for async frames when flushing
This was causing incorrect output when seeking, especially
when used with a multithreaded source like `videotestsrc n-threads=2`.
It should now correctly wait for frames still being processed by VT
while vtdec is flushing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3922>
2023-02-10 18:35:40 +00:00
Alessandro Bono
fcd983440e win32ipcutils: Add missing include
We are using std::isspace() with one parameter. That function is defined
in the cctype header.

```
win32ipcutils.cpp(34): error C2672: 'std::isspace': no matching overloaded function found
win32ipcutils.cpp(34): error C2780: 'bool std::isspace(_Elem,const std::locale &)': expects 2 arguments - 1 provided
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3933>
2023-02-10 14:35:20 +00:00
Thibault Saunier
0f577533e6 examples: Add an option to disable tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3930>
2023-02-10 12:59:55 +00:00
Sebastian Dröge
fc5bad5f75 examples: webrtc: rust: Fix a couple of minor clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3928>
2023-02-10 11:43:00 +00:00
Sebastian Dröge
28ab612a88 examples: webrtc: rust: Update to gstreamer-rs 0.20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3928>
2023-02-10 11:43:00 +00:00
Edward Hervey
f072b25940 adaptivedemux2: Use track ID for debugging
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3890>
2023-02-10 10:56:52 +00:00
Edward Hervey
5e193730db adaptivedemux2: Split track id from event stream-id
The id is used for naming of the various objects and debugging. We don't
want/need it to be obfuscated with the massive upstream id.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3890>
2023-02-10 10:56:52 +00:00
Tim-Philipp Müller
edaf3252c5 tests: mark elements_srtp.test_play test as flaky
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1777

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3927>
2023-02-10 10:04:39 +00:00
Edward Hervey
edc66ce003 mxfdemux: Add support for FFV1 demuxing
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3899>
2023-02-10 10:02:25 +01:00
Edward Hervey
3768ef89b9 mxf: simplify essence track matching
By directly providing a valid descriptor to match against

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3899>
2023-02-10 10:02:25 +01:00
Edward Hervey
2d5b71d6e3 mxfdemux: Handle empty tracks
This can happen when tracks aren't fully/properly detected due to new descriptors

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3899>
2023-02-10 10:02:25 +01:00
Edward Hervey
206ed12368 mxfdemux: Handle sub-descriptor in generic descriptors
Specificied in S377-1 (2019) B.2 Generic Descriptor

This is one of the ways to store more "sub descriptors" for a single track

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3899>
2023-02-10 10:02:25 +01:00
Edward Hervey
56a50c3008 mxf: Add convenience function for looking up primer tags
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3899>
2023-02-10 06:25:29 +01:00
Jan Alexander Steffens (heftig)
aeb262a7e1 pad: Don't leak user_data in gst_pad_start_task
When the task already exists, we forgot to free the passed `user_data`.
This wasn't an issue for most C code, which doesn't pass a
`GDestroyNotify`, but bindings such as gstreamer-rs do!

That said, allocating a trampoline in gstreamer-rs just for it to get
thrown away again is awkward. Maybe we need a `gst_pad_resume_task`?

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3920>
2023-02-09 15:40:56 +00:00
Sebastian Dröge
5119ca25e4 av1parser: Don't consider unknown metadata OBUs a bitstream error
Just don't parse them.

Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1774

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3898>
2023-02-09 14:52:58 +00:00
Sebastian Dröge
5486ed24a5 qtmux: Implement writing of av1C version 1 box
Version 0 is ancient and not specified in any documents. Take it
directly from the `codec_data` if presents or otherwise try to construct
a reasonably looking `av1C` box.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3882>
2023-02-09 14:04:06 +00:00
Sebastian Dröge
8593a58916 qtdemux: Drop av1C version 0 parsing and implement version 1 parsing
The av1C box is optional so dropping parsing does not break anything
fundamentally, and there seems to be no historical record how version 0
even looks like while the comments and the parsing disagreed with each
other.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3882>
2023-02-09 14:04:06 +00:00
Edward Hervey
3d506a0978 subparse: Properly forward segment seqnum
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3913>
2023-02-08 14:12:07 +00:00
Thibault Saunier
1bab0ef2aa bin: Do not deactivate pad in NULL_TO_READY
Since b76d336549
pads are deactivated when going to READY but in `uridecodebin(3)`, the
sources source pads are activated while in NULL state (when PULL mode is
supported), meaning that we are ending up deactivating those pads in
NULL_TO_READY, breaking the pipeline.

The intent of the commit mentioned above is to ensure that the pads are
deactivated either in PAUSED_TO_READY or READY_TO_READY, so it should
be safe to avoid deactivating in NULL_TO_READY.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3849>
2023-02-08 13:23:46 +00:00
Patricia Muscalu
c3e52d5c4f rtph264pay: Don't insert SPS/PPS before the second image slice
Only the first slice, for which fist_mb_in_slice is set to 0,
should trigger insertion of SPS and PPS buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3402>
2023-02-08 12:10:11 +00:00
Thibault Saunier
4067bbdd91 uridecodebin: Set source element to READY before querying it
Generating the source element is done when uridecodebin is doing the
READY to PAUSED state change, so it is reasonable to set the new source
element to that state.

This also allows detecting early failures with backing libraries or
hardware (checks done in NULL->READY).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3857>
2023-02-08 11:07:21 +00:00
Philipp Zabel
485c8ef4b5 gleglimage: cache EGL images per DmabufUpload
Do not store cached EGL images in GstMemory QData. Instead, use a
per-DmabufUpload GHashTable to store cache entries with a weak
reference to the GstMemory.

This allows two glupload elements on separate tee branches to have
their own EGL image cache. For this pipeline:

  gst-launch-1.0 v4l2src ! tee name=t \
      t. ! queue ! glupload ! fakesink
      t. ! queue ! glupload ! fakesink

this gets rid of the occasional critical error message:

  GStreamer-CRITICAL **: 08:26:33.194: gst_mini_object_unref: assertion 'GST_MINI_OBJECT_REFCOUNT_VALUE (mini_object) > 0' failed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3880>
2023-02-07 19:32:14 +00:00
medithe
3943503fc1 gstreamer: bin: Don't unlock unlocked mutex in gst_bin_remove_func()
Calling `g_mutex_unlock(mutex)` leads to an undefined behavior if the
mutex is not locked by the current thread.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3903>
2023-02-07 19:03:43 +02:00
Edward Hervey
d10f9a00e2 closedcaption: Don't leak caps event
All events that we handle should be unreffed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3902>
2023-02-07 08:51:18 +01:00
Jan Schmidt
621604aa3e webrtc: Calculate the jitter for remote-inbound-rtp stats
Populate the clock-rate in the internal stats structure, so
it can be used by the _get_stats_from_remote_rtp_source_stats()
method to calculate remote receivers' jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3900>
2023-02-07 04:58:04 +11:00
Jan Schmidt
615a019457 webrtcbin: Report full codec-stats for source pads
Use the current caps for webrtcbin srcpads, as received_caps
are only stored for sink pads based on incoming caps events.

Makes it so that webrtcbin stats reports contain fuller
codec information.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3900>
2023-02-07 04:49:34 +11:00
Anders Hellerup Madsen
acb8f2ee5d glstereosplit: use gst_display_ensure_context
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3843>
2023-02-06 16:16:14 +01:00
Anders Hellerup Madsen
f0040149a0 glbasefilter: use gst_display_ensure_context
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3843>
2023-02-06 16:15:46 +01:00
Anders Hellerup Madsen
ecd9a4e37c glbasemixer: use gst_display_ensure_context
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3843>
2023-02-06 16:15:06 +01:00
Anders Hellerup Madsen
7bee4619dd glbasesrc: use gst_display_ensure_context
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3843>
2023-02-06 16:14:34 +01:00
Anders Hellerup Madsen
0da0da69aa gldisplay: Add gst_gl_display_ensure_context
See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/439

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3843>
2023-02-06 16:14:07 +01:00
Adrian Fiergolski
06b778e0a1 avtp: specify the required version of libavtp
Support of RVF requires libavtp in version 0.2.0 at least.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3897>
2023-02-06 13:31:51 +00:00
Enrique Ocaña González
92a4cfe20f qtdemux: Don't emit GstSegment correcting start time when in MSE mode
When using qtdemux in a pipeline that should only work as a pure demuxer (not
for actual playback), qtdemux shouldn't emit new GstSegments to correct
the start time (jump to the future) to ensure that the user experiences no
playback delay. By doing so, it's generating the wrong segments when an append
of data from the past happens. When that happens, downstream elements such as
parsers (eg: aacparse) may clip those buffers laying before the GstSegment and
create problems on the GStreamer client app (eg: WebKit).

Getting buffers clipped out because of the wrong GstSegments started becoming
a problen when this commit was introduced:

ab6e49e9cc audioparsers: add back segment clipping to parsers that have lost it

This clipping makes test DASH shaka 35 from MVT tests[1] to fail in
WebKitGTK/WPE (at least) and can potentially cause a number of other problems
in the WebKit Media Source Extensions (MSE) code.

Note that this new behaviour of not emitting new GstSegments only makes sense
when qtdemux is being used as a pure demuxer and not as part of a regular
pipeline. This is why the variant field has been added. When equal to
VARIANT_MSE_BYTESTREAM, it will make qtdemux behave differently in push mode,
taking decisions that meet the expectations for an MSE-like processing mode.
This kind of tweaks have been done in the past for MSS streams, for instance.
That code has been refactored to use VARIANT_MSS_FRAGMENTED now, instead of
its own dedicated boolean flag.

Co-authored by: Alicia Boya García <ntrrgc@gmail.com>

...who suggested to use "variant: mse-bytestream" in the caps to identify that
mode, as proposed in her unmerged patch:

https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/467

[1] https://github.com/rdkcentral/mvt

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3867>
2023-02-06 12:42:49 +00:00
Nirbheek Chauhan
77b8547586 meson: Allow sysdeps to be forced as fallback subprojects
The original code was too complicated; likely created before the
provide section existed for wraps:

https://mesonbuild.com/Wrap-dependency-system-manual.html#provide-section

Now you can do --force-fallback-for=pygobject and it'll actually work.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3804>
2023-02-06 09:26:02 +00:00
Ma, Mingyang
99cdc3a965 msdkenc: Let runtime decide parameters
Some parameters can be determined by runtime instead of default values. So unset the default and let runtime choose the best parameters

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3660>
2023-02-06 02:24:54 +00:00
Nirbheek Chauhan
033a71e405 webrtc examples: Use webrtc.gstreamer.net
Actually just a CNAME to webrtc.nirbheek.in for now, but it allows
replacement / hosting without my involvement, so reduces the bus
factor.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3802>
2023-02-04 13:37:02 +00:00
Sebastian Dröge
a0ccb6b558 svtav1enc: Use G_DECLARE_FINAL_TYPE and GST_ELEMENT_REGISTER_DEFINE 2023-02-03 22:14:18 +02:00
Sebastian Dröge
aca2bad25c svtav1enc: Fix compilation with SVT-AV1 1.1 and drop GStreamer 1.16 compatibility 2023-02-03 22:14:18 +02:00
Sebastian Dröge
5bc92375c9 svtav1enc: Fix indentation 2023-02-03 22:14:18 +02:00
Sebastian Dröge
7890a1f8c7 svtav1: Integrate into the build system properly 2023-02-03 22:14:18 +02:00
Sebastian Dröge
b15efacf84 svtav1: Merge SVT-AV1 encoder into gst-plugins-bad
This is based on d5e1e2a586020854733f6b0806064d0c900c88d2 from
https://gitlab.com/AOMediaCodec/SVT-AV1.
2023-02-03 22:13:30 +02:00
Sebastian Dröge
716aaa562b net: ptp: Use GSubprocess instead of lower-level GLib APIs that don't work on Windows
libgstnet depends on GIO already anyway so we can as well make use of it
instead of a half-baked Windows implementation that doesn't actually
work.

As a next step, the helper process also needs to be made usable on
Windows.

See https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/1259

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3887>
2023-02-03 18:51:17 +00:00
Tim-Philipp Müller
85476eab08 kms: drop use of GSlice allocator and remove unnecessary check
g_new0() will never return NULL but just abort if it can't
allocate memory (same for g_slice_new).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:10 +00:00
Tim-Philipp Müller
35405de344 shm: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:10 +00:00
Tim-Philipp Müller
3f94d7ec37 midiparse: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:10 +00:00
Tim-Philipp Müller
d95d3e39af cc708overlay: bump pango requirement and drop no longer required locking
Gets rid of GSlice allocation that's never freed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:10 +00:00
Tim-Philipp Müller
7679011d1d validate: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:10 +00:00
Tim-Philipp Müller
cae6c6c73a gst-omx: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:10 +00:00
Tim-Philipp Müller
0d9bdf238c gst-docs: drop use of GSlice in example code
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:09 +00:00
Tim-Philipp Müller
18a3c32323 ges: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:09 +00:00
Tim-Philipp Müller
f8817a8e8d ges: nle: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:09 +00:00
Tim-Philipp Müller
06e9d78ade gst-examples: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:09 +00:00
Tim-Philipp Müller
8a047a619e gst-libav: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:09 +00:00
Tim-Philipp Müller
f5977dae15 rtsp-server: drop use of GSlice allocator
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3784>
2023-02-03 17:48:09 +00:00
Edward Hervey
0639f117cb hlsdemux2: Remove enable-llhls property
This was only used for testing purposes

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:24 +00:00
Edward Hervey
854683c871 hlsdemux2: Don't leak PDT datetime
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:24 +00:00
Edward Hervey
96613c45fb adaptivedemux2: Don't leak taglist
Clarify the ownership in the documentation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:24 +00:00
Edward Hervey
123030feac adaptivedemux2: Don't leak track tags
The tags are fully transfered to this function

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
6f6c0cbbaf adaptivedemux2: Log request duration in debug output
When completing, log how long a HTTP request took into the debug output.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
714628f1ec hlsdemux2: Improve live playlist update intervals
The live playlists should be updated at a defined interval. The problem is that
this interval was used *after* the playlist was finally received and processed,
which resulted in a gradual shift happening in playlist updates.

Instead store and use the time at which playlists were requested to determine
when the next one should be downloaded.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
6684aee14c hlsdemux2: Fix playlist reload interval when unchanged
When falling back to using the regular last segment, use that duration as the
identical-playlist reload interval (and not the playlist target duration which
could be much larger)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
5935c8049a hlsdemux2: Fix position searching
The scanning is done in a reverse order, the proper full checks to do are
therefore:
* If the position is beyond half a "segment duration", it's in the following
segment
* If the position is within the first half of a segment, it's in that one
* If the segment is the first one and the position is within half a duration
backwards, we consider the position as being within that first segment

Also handle the case where a "partial only" segment doesn't have a reliable
duration, and therefore use the playlist target duration instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
1c6364673d hlsdemux2: Handle all cases for starting segment calculation
The implementation wouldn't work with regular HLS streams (i.e. the final
fallback).

Now that the implementation uses time to search for the starting
segment (instead of just the n-th from the end), we can specify the correct
hold_back fallback value from the RFC

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
3129970c8a hlsdemux2: Fix buffering threshold calculation and handling
* The checks for smaller values were wrong
* Properly initialize the stream default recommended buffering threshold so that
  a default (10s) value is used until the subclass can provide a proper value

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Edward Hervey
eb1eb64506 hlsdemux2: Make sure simple media playlist is properly primed
By setting/propagating stream time initially

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
3d0e8aa07e adaptivedemux2: Fix manifest access during seeking query
Avoid a deadlock if a downstream seeking query happens while the scheduler
thread is holding the manifest lock (for example during a seek back to live).

Instead, do a more elaborate fix where the external calls that need access to a
'manifest' access a copy that's updated during a manually triggered manifest
update callback.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
5334007a0b adaptivedemux2: Symbol hygiene cleanup
Rename track_dequeue_data_locked() to
gst_adaptive_demux_track_dequeue_data_locked(), since it's non-static.

Make find_stream_for_track_locked() static since it's only used in the main
gstadaptivedemux.c file.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00