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Author SHA1 Message Date
Maksym Khomenko
e5fd2c3568 webrtcsrc: add turn-servers property 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1380>
 2023-11-04 10:19:45 +00:00
Mathieu Duponchelle
5371eb52ad Port to AWS SDK 0.57/0.35 
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
 2023-11-03 15:13:45 +00:00
Sebastian Dröge
f7745a336f aws: Update to test-with 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
 2023-11-03 15:13:45 +00:00
Sebastian Dröge
a33f29365a sccparse: Fix leading spaces between the tab and caption data 
CCExtractor is creating files like this.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1378>
 2023-11-02 21:59:02 +02:00
Sebastian Dröge
16b917abb1 Update for gst::Rank API changes 2023-11-02 14:10:59 +02:00
Piotr Brzeziński
436b6d8efb gstwebrtc-api: Patch webrtc-adapter to fix Safari behaviour 
There's currently a Safari-side bug causing webrtc-adapter to be unable to correctly shim the empty-candidate scenario
which we're using. This patch is very much a workaround and should be removed as soon as Safari and/or webrtc-adapter
fixes this on their side.

https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/439
https://github.com/webrtcHacks/adapter/issues/1140

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1377>
 2023-10-30 16:36:11 +00:00
Sebastian Dröge
16c00ae3f5 Set sync=false in rsfilesink / s3sink 
BaseSink defaults to sync=true and that doesn't make much sense for
these elements.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1376>
 2023-10-30 17:38:46 +02:00
Sebastian Dröge
855b03a9ea Use let-else instead of match for weak reference upgrades 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1375>
 2023-10-30 11:34:35 +02:00
Sebastian Dröge
74c04d79c9 gtk4: Use async-channel instead of the glib MainContext channel 
The latter will be removed in favour of using async code in the future,
and async code generally allows for more flexible message handling than the
callback based MainContext channel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1375>
 2023-10-30 11:21:25 +02:00
Sebastian Dröge
b771afe8be deny: Update duplicated dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1374>
 2023-10-27 10:20:54 +03:00
Sebastian Dröge
557b249e11 Update to AWS SDK 0.34 and tracing-log 0.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1374>
 2023-10-27 10:19:15 +03:00
Jan Alexander Steffens (heftig)
e3e58ac0be livesync: Remove the stop from outgoing segments 
Our buffer duplication can extend a segment indefinitely.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/452
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1372>
 2023-10-25 19:34:47 +02:00
Jan Alexander Steffens (heftig)
f1ba498b52 livesync: Keep existing buffer duration in some cases 
Resize a repeat buffer only if caps gave us a duration to use, or we
consider its current duration unreasonable.

In particular, for audio streams we should prefer reusing the buffer
size upstream gave us, as we did before 6633cc4046.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1372>
 2023-10-25 19:34:47 +02:00
Jan Alexander Steffens (heftig)
59beade079 livesync: Split fallback_duration into in_ and out_duration 
Make it independent of the `latency`; this was inconsistent anyway,
where the default latency of zero got you a fallback duration of 100 ms
and something else got you half the latency.

Maintain a separate duration for the `in` and the `out` side so we
change the duration of repeat buffers after a caps change, not just
before.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1372>
 2023-10-25 19:08:16 +02:00
Guillaume Desmottes
f94ecfc7a6 livesync: display jitter when waiting on clock 
We already log the result of the clock wait call so may as well log the
returned jitter.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1370>
 2023-10-25 14:26:19 +02:00
Guillaume Desmottes
13dae0f0d0 livesync: log new pending segments 
The debug print of the event does not display details about the segment:
  Unqueueing Some(Event(Event { ptr: 0x7fa3e0002580, type: "segment", seqnum: Seqnum(479), structure: Some(GstEventSegment { segment: (GstSegment) ((GstSegment*) 0x7fa3e8001d00) }) }))

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1370>
 2023-10-25 14:24:35 +02:00
Jan Alexander Steffens (heftig)
ee93448de7 livesync: example: Add identities single-segment=1 
These let us change the runtime offset of the test buffers via pad
offsets without pushing new segments into livesync, which is necessary
to demo the late-threshold behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 12:55:06 +02:00
Jan Alexander Steffens (heftig)
6633cc4046 livesync: Use fallback_duration for audio repeat buffers as well 
Don't depend on upstream giving us sanely-sized buffers if we want to
repeat.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 12:55:06 +02:00
Jan Alexander Steffens (heftig)
4ac7d0415b livesync: Separate out_buffer duplicate status from GAP flag 
Otherwise we might get confused by upstream GAP buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 12:10:40 +02:00
Jan Alexander Steffens (heftig)
2f36bd5d77 livesync: Handle flags and late buffer patching after queueing 
This makes the chain function almost independent of the output state. We
still do the early discard check with `buffer_is_backwards` so we don't
try to queue buffers we can't use, allowing us to fast-forward upstream
without blocking on the src task.

Don't accept `LateOverThreshold` buffers when we have `pending_caps` or
a `pending_segment`. We need to apply these first before we can sensibly
patch buffers from the new stream.

Deduplicate most of the output buffer patching code into a new
`patch_output_buffer` method.

For: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/450
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:41 +02:00
Jan Alexander Steffens (heftig)
7c48a299c3 livesync: Simplify num_duplicate counting 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
17a2448237 livesync: Move num_in counting to the src task 
This is in preparation for moving more accept/discard logic to the src
task, so we can only count `num_in` here.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
1740a8e363 livesync: Move a notify closer to the interesting state change 
Move the `notify_all` to where we pop the buffer. We're moving within a
single state lock so no change in behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
44f2195674 livesync: Replace an if-let with match 
No change in behavior, yet. Separate commit to ease reviewing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
62791bfb47 livesync: Clean up state handling 
- Separate resetting state more cleanly, introducing `set_flushing`,
  `sink_reset` and `src_reset`.
- Clear the queue early when we flush, in order to unblock waits on
  query responses.
- Return an error when we fail to start, pause or stop the task.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
d663f708ef livesync: Log a category error when we are missing the segment 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
6567041a3d livesync: Improve audio duration fixups 
- An entirely missing duration is now only logged at debug level instead
  of pretending the duration was zero and warning about it.
- Silently fix up a duration difference up to one sample.
- Error when we fail to calculate the duration; don't try to apply the
  `fallback_duration` to a non-video stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:52:40 +02:00
Jan Alexander Steffens (heftig)
0a45f776e0 livesync: Simplify start_src_task and src_loop 
This should effect no change in behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:03:15 +02:00
Jan Alexander Steffens (heftig)
01386b8451 livesync: Rename activatemode methods to *_activatemode 
This matches the other plugins.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1369>
 2023-10-25 11:03:14 +02:00
Arun Raghavan
d27a04e067 hlssink3: Close the playlist giostreamsink on stop if possible 
This is a property that will be available from GStreamer 1.24, and will
ensure that we are able to flush the playlist during the READY->NULL
transition instead of when the element is freed.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/423
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1360>
 2023-10-24 21:03:14 +00:00
Arun Raghavan
a49a5dcb11 s3: tests: Remove emoji-based tests for now 
These break hotdoc, which we need to fix first.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
 2023-10-24 12:52:12 -04:00
Arun Raghavan
bb26e04a55 aws: s3: Properly percent-decode GstS3Url 
We previously only percent-decoded the first fragment. This doesn't
necessarily harm anything, but for consistency we keep the structure
un-encoded, and encode when converting to a string representation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
 2023-10-24 12:52:12 -04:00
Arun Raghavan
51129febeb aws: s3sink: Fix handling of special characters in key 
Properly URL-encode the string if needed, and add some tests for a
couple of cases.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/431
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1333>
 2023-10-24 12:52:12 -04:00
Sebastian Dröge
829469d0fe rtpav1depay: Don't push stale temporal delimiters downstream 
Only push them downstream once a complete OBU was assembled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
 2023-10-24 11:13:35 +00:00
Sebastian Dröge
1f5e9a9335 rtpav1depay: Skip unexpected leading fragments 
If a packet is starting with a leading fragment but we do not expect to
receive one, then skip over it to the next OBU.

Not doing so would cause parsing of the middle of an OBU, which would
most likely fail and cause unnecessary warning messages about a
corrupted stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
 2023-10-24 11:13:35 +00:00
Sebastian Dröge
b1894a0acb deny: Remove unnecessary ignore entry 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1368>
 2023-10-24 09:55:54 +03:00
Sebastian Dröge
73ff822d24 Update to quick-xml 0.31 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1368>
 2023-10-24 09:55:50 +03:00
Jordan Petridis
a2d7f42138 Fix compilation after glib bindings changes 
loggable_error! can now expand variables and we no longer need
the format! on our side.

https://github.com/gtk-rs/gtk-rs-core/pull/1210

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1366>
 2023-10-22 01:20:56 +03:00
Nirbheek Chauhan
895ee1d00b ci: Make meson warnings fatal 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1364>
 2023-10-18 18:49:02 +05:30
Nirbheek Chauhan
235e609eff meson: Bump requirement to 1.1 
WARNING: Project specifies a minimum meson_version '>= 0.60' but uses features which were added in newer versions:
* 1.1.0: {'feature_option.enable_if()'}

Caused by https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1363

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1364>
 2023-10-18 18:07:50 +05:30
Nirbheek Chauhan
84b0dd8980 meson: Enable the RTP option when WebRTC is enabled 
And make the webrtc option yielding, see:

https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5505

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1363>
 2023-10-18 14:47:16 +05:30
Nirbheek Chauhan
c4a788b97b .gitignore: Ignore the meson subproject wrap hash 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1363>
 2023-10-18 14:47:16 +05:30
Olivier Crête
a946895fad Revert "deny: Temporarily allow a duplicated tungstenite dependency" 
LiveKit has now been fixed.

This reverts commit 23e1bfa720.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1293>
 2023-10-18 10:31:14 +03:00
Sebastian Dröge
2ce04c6a78 webrtc: Update to livekit 0.2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1293>
 2023-10-18 10:30:59 +03:00
Sebastian Dröge
aacfe546d5 deny: Update for duplicated redox_syscall dependency 2023-10-18 10:25:27 +03:00
Sebastian Dröge
d468e1e4a6 Clean up usage of pad probes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1361>
 2023-10-17 08:44:06 +03:00
François Laignel
50dd519c4f net/webrtcsrc: define signaller property as CONSTRUCT_ONLY 
The "signaller" property used to be defined as MUTABLE_READY which meant that
the property was always set after `constructed()` was called.

Since `connect_signaller()` was called from `constructed()`, only the default
signaller was used.

This commit sets the "signaller" property as CONSTRUCT_ONLY. Using a builder,
this property will now be set before the call to `constructed()`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1324>
 2023-10-12 17:38:09 +00:00
François Laignel
785c9557c8 net/webrtcsink: drop State lock before calling set-local-description 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1325>
 2023-10-12 15:45:58 +00:00
François Laignel
c021e2b69f net/webrtcsink: don't miss ice candidates 
During `on_remote_description_set()` processing, current session is removed
from the sessions `HashMap`. If an ice candidate is submitted to `handle_ice()`
by that time, the session can't be found and the candidate is ignored.

This commit wraps the Session in the sessions `HashMap` so an entry is kept
while `on_remote_description_set()` is running. Incoming candidates received by
`handle_ice()` will be processed immediately or enqueued and handled when the
session is restored by `on_remote_description_set()`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1325>
 2023-10-12 15:45:58 +00:00
Sebastian Dröge
42008fb895 aws: Update to test-with 0.11 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1358>
 2023-10-12 06:57:28 +00:00