Commit graph

531 commits

Author SHA1 Message Date
Mathieu Duponchelle
6747b11cb9 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/1388>
2023-11-10 16:05:12 +02:00
Sebastian Dröge
aa491e0d54 aws: Update to test-with 0.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 16:05:06 +02:00
Sebastian Dröge
2bec8a8eaa 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/1388>
2023-11-10 16:04:46 +02:00
Sebastian Dröge
1ceaf8b179 Use let-else instead of match for weak reference upgrades
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 16:04:38 +02:00
Sebastian Dröge
e501ee0340 Update to AWS SDK 0.34 and tracing-log 0.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 16:02:46 +02:00
Arun Raghavan
54bc4012d2 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/1388>
2023-11-10 16:00:31 +02:00
Seungha Yang
4bb82748b9 hlssink3: Various cleanup
* Simplify state/playlist management
* Fix a bug that segment is not deleted if location contains directory
and playlist-root is unset
* Split playlist update routine into two steps, adding segment
to playlist and playlist write

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:59:17 +02:00
Seungha Yang
9b229eb456 hlssink3: Don't remove old files if max-files is zero
Follow hlssink2 element's behavior

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:59:13 +02:00
Seungha Yang
c41e4c0304 hlssink3: Remove unused deps
gstreamer-base dep is unused. And use gst::glib

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:59:04 +02:00
Seungha Yang
1e5f499b3f hlssink3: Use Path API for getting file name
Current implementation does not support Windows path separator.
Use Path API instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:58:42 +02:00
Seungha Yang
d5e6ca3e5a hlssink3: Use sprintf for segment name formatting
The zero-padded naming requirement is unnecessary. Use simple
sprintf instead

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:58:37 +02:00
Arun Raghavan
cf4d7c5431 hlssink3: Minor PDT-related naming fixups
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:58:32 +02:00
rajneeshksoni
12e26cef8b hlssink3: Add property track-pipeline-clock-for-pdt.
This is required to take care of clock skew between
system time and pipeline time.
`track-pipeline-clock-for-pdt: true` mean utd time is
sampled for first segment and for subsequent segments
keep adding the time based on pipeline clock. difference
of segment duration and PDT time will match.
track-pipeline-clock-for-pdt: false` mean utd time is
sampled for each segment. system time may jump forward
or backward based on adjustments. If application needs
to synchronization of external events `false` is
recommended.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:58:27 +02:00
rajneeshksoni
4971084f63 hlssink3: Allow adding EXT-X-PROGRAM-DATE-TIME tag.
- connect to `format-location-full` it provide the first
sample of the fragment. preserve the running-time of the
first sample in fragment.
- on fragment-close message, find the mapping of running-time
to UTC time.
- on each subsequent fragment, calculate the offset of the
running-time with first fragment and add offset to base
utc time

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:58:22 +02:00
Arun Raghavan
12418a3f52 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/1388>
2023-11-10 15:56:11 +02:00
Arun Raghavan
dd05c0d51a 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/1388>
2023-11-10 15:56:06 +02:00
Sebastian Dröge
5160b6c30f 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/1388>
2023-11-10 15:55:58 +02:00
Sebastian Dröge
3d3dc19fc2 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/1388>
2023-11-10 15:55:51 +02:00
Sebastian Dröge
bf48e76b59 webrtc: Update to livekit 0.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:54:20 +02:00
François Laignel
9fd9f0eb83 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/1388>
2023-11-10 15:51:32 +02:00
François Laignel
ff4b1cb757 net/webrtcsink: drop State lock before calling set-local-description
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:51:15 +02:00
François Laignel
6dc624ac2f 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/1388>
2023-11-10 15:51:10 +02:00
Sebastian Dröge
f5d6d703c5 aws: Update to test-with 0.11
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:51:03 +02:00
Lieven Paulissen
56095c5434 ndisrc: Assume input with more than 8 raw audio channels is unpositioned
gst_audio_channel_positions_from_mask() will otherwise print warnings
all the time.

Fixes #444

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:50:59 +02:00
Maksym Khomenko
2c923128dc webrtcsrc: use @watch instead of @to-owned
@to-owned increases refcount of the element, which prevents the object from proper destruction, as the initial refcount with ElementFactory::make is larger than 1.

Instead, use @watch to create a weak reference and unbind the closure automatically if the object gets destroyed

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:50:53 +02:00
Sebastian Dröge
36273c0eae Update to AWS SDK 0.33
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:50:47 +02:00
Taruntej Kanakamalla
7bfb86c6cc net/webrtc/whip_signaller: Use the correct URL during redirect
Copy of 90e06dc3 for whipclientsink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:50:42 +02:00
Maksym Khomenko
aabbe49dc3 webrtcsink: README: add documentation for custom signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:50:36 +02:00
Maksym Khomenko
569dfadcab webrtcsink: add custom signaller example
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:50:29 +02:00
Sebastian Dröge
107c610bb0 Update to AWS SDK 0.32
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1388>
2023-11-10 15:50:23 +02:00
Sebastian Dröge
2b8728103f Update versions to 0.11.1 2023-10-04 23:35:35 +03:00
Sebastian Dröge
b9e8f4cbb3 ndi: Comment out empty Opus handling and add FIXME comment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
2023-10-04 19:30:49 +03:00
Sebastian Dröge
4eeff19075 Fix indentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
2023-10-04 19:19:23 +03:00
Sebastian Dröge
b360b5247e Update gst-plugin-version-helper version requirement to 0.8
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
2023-10-04 19:15:53 +03:00
Sebastian Dröge
29697dae45 Fix various new 1.73 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
2023-10-04 19:10:02 +03:00
Piotr Brzeziński
e792fdbe48 webrtc: Fix paths in README
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
2023-10-04 19:07:14 +03:00
Sean DuBois
4a1a7864cf net: webrtc/webrtchttp: Respect HTTP redirects
Properly follow redirect URL. Before new request would be made, but with
original URL again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
2023-10-04 19:06:03 +03:00
Sebastian Dröge
a168456428 Update to AWS SDK 0.31
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1350>
2023-10-04 19:04:21 +03:00
Sebastian Dröge
9187d0e11e onvifmetadataparse: Skip metadata frames with unrepresentable UTC time
Previously we would panic, which causes the element to post an error
message. Instead, simply skip metadata frames if their UTC time since
the UNIX epoch can't be represented as nanoseconds in u64.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
2023-09-20 19:58:43 +03:00
Seungha Yang
468f73092d webrtcsink: Propagate GstContext messages
Implement CustomBusStream so that NEED_CONTEXT and HAVE_CONTEXT
messages from session/discovery can be forwarded to parent
pipeline and also GstContext can be shared.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
2023-09-20 19:58:13 +03:00
Seungha Yang
2b1d07a757 webrtcsink: Add support for d3d11 memory and qsvh264enc
Adding d3d11 memory and qsvh264enc support

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
2023-09-20 19:58:06 +03:00
Mathieu Duponchelle
33156b52c8 webrtcsink: fix codec selection discoveries
Since ab1ec12698:

webrtcsink: Add support for pre encoded streams

Discovery pipelines for remote offers were no longer fed any buffers.

While some encoders could already produce caps with no input buffers,
others, such as x264enc, simply hung forever. This resulted in no answer
getting produced if for instance video-caps were constrained to H264.

Fix this by tracking discovery pipelines at the State rather than the
InputStream level, removing the useless distinction of Initial vs.
CodecSelection discoveries, and always feeding all the current
discovery pipelines with incoming buffers.

For reference, the issue here was that codec selection discoveries were
assigned to local clones of InputStreams, not tracked anywhere, and thus
not iterated for discoveries when queuing incoming buffers from the
chain function, as it only looked at the original instance of
InputStream's in state.streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
2023-09-20 19:57:57 +03:00
Robert Ayrapetyan
391ba3f5f8 webrtcsink: fix TWCC extension adding
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
2023-09-20 19:57:40 +03:00
Sebastian Dröge
dc1e45eaa1 Update to AWS SDK 0.30
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1331>
2023-09-20 19:57:00 +03:00
Sebastian Dröge
f35603a020 webrtc: Add versions to gst_plugin_webrtc_protocol dependency 2023-08-10 16:57:11 +03:00
Sebastian Dröge
982b5d6ef9 Add version to the gst-plugin-version-helper dependency 2023-08-10 16:56:04 +03:00
Sebastian Dröge
67c3732b94 Update dependencies to release branches of gtk-rs / gtk4-rs / gstreamer-rs 2023-08-09 18:14:18 +03:00
Sebastian Dröge
9b1853c1de Update versions to 0.11.0 2023-08-09 18:04:38 +03:00
Sebastian Dröge
3b41f206bc Don't generate .def files for plugins
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
2023-08-09 13:54:34 +03:00
Sebastian Dröge
b3826c108d webrtc: Update to async-tungstenite 0.23
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
2023-08-09 13:18:44 +03:00