Commit graph

116531 commits

Author SHA1 Message Date
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
Seungha Yang
9fa5fbc25e gitignore: Ignore gtest.wrap
Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3893>
2023-02-05 04:28:00 +09:00
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
Jan Schmidt
6bb74ed2a0 adaptivedemux2: Fix download error handling more
gst_adaptive_demux2_stream_finish_download() will already schedule another
fragment download if it can so don't fall through to the retry code that will
also try and schedule a download (triggering an assert).

Fix the logic in general to retry advancing into the live seek range once.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
b1354058e1 hlsdemux2: Immediately request playlist after URI changes
When the stream switches to a new playlist / variant while the loader is waiting
on a timer to refresh the old playlist, cancel the timer and submit the request
for the new URI.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
6d7d3d93e6 hlsdemux2: Re-add support for fallback variant URLs
fallback variant URLs get accumulated into a list in the variant now. If there's
one available, switch to it after a variant update failure (failure to load the
variant 3 times)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
d5b8929315 hlsdemux2: Demote log message
Don't complain loudly about replacing the current pending playlist, just log it
at debug level

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
91c8f3f990 hlsdemux2: Wait for playlist load after a switch
Check in update_fragment_info() if the playlist we want has actually been loaded
yet, and return BUSY if not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3883>
2023-02-03 16:52:23 +00:00
Jan Schmidt
2b93dae59a hlsdemux2: Handle async playlist loading failures
Add failed variant playlists to a list and failover to other variants until
there is none left

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