Commit graph

521 commits

Author SHA1 Message Date
Maksym Khomenko
a9719cada2 webrtcsink: add custom signaller example
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
2023-10-06 12:58:03 +03:00
Sebastian Dröge
1c4833bc5d Update to AWS SDK 0.32
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1352>
2023-10-06 09:11:17 +03:00
Sebastian Dröge
4569b7eca6 Fix various new 1.73 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1347>
2023-10-03 17:47:30 +03:00
Sebastian Dröge
450ffbe452 Update for VideoFrame / GLVideoFrame API changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1345>
2023-10-02 13:25:25 +03:00
Piotr Brzeziński
fe4273ca2a webrtc: Fix paths in README
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1342>
2023-09-29 17:05:29 +02:00
Sean DuBois
90e06dc37b 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/1335>
2023-09-26 19:29:41 -04:00
Seungha Yang
22cc8c4986 hlssink3: Update README
Mention newly added hlscmafsink element and new properties

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:34:05 +09:00
Seungha Yang
1888a2eb82 hlscmafsink: Add live recording example
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:34:05 +09:00
Seungha Yang
52117e4b11 hlsbasesink: Add enable-endlist property
Write "EXT-X-ENDLIST" tag at the end of stream if enabled, and
default to "TRUE" which is the hlssink2's behavior as well

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:34:05 +09:00
Seungha Yang
7835d78b3d hlssink3: Add hlscmafsink element
Adding cmafmux based hls sink element

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:34:00 +09:00
Seungha Yang
5b563006f9 hlssink3: Add baseclass implementation
Adding HlsBaseSink class to make code reusable

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:32:16 +09:00
Seungha Yang
0fe69cea9f 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/1306>
2023-09-25 21:32:16 +09:00
Seungha Yang
d8546dd140 hlssink3: Don't remove uri from playlist if playlist-length is zero
Behave as documented in property description

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:32:16 +09:00
Seungha Yang
8e4863e9cd 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/1306>
2023-09-25 21:32:16 +09:00
Seungha Yang
a8d67cc607 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/1306>
2023-09-25 21:32:16 +09:00
Seungha Yang
c4d371d163 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/1306>
2023-09-25 21:32:16 +09:00
Seungha Yang
7f16ac3915 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/1306>
2023-09-25 21:32:16 +09:00
Sebastian Dröge
9595c6a1e5 Update to AWS SDK 0.31
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1334>
2023-09-25 13:36:12 +03:00
Arun Raghavan
8bbfb10cba hlssink3: Minor PDT-related naming fixups
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1332>
2023-09-20 16:52:55 -04:00
rajneeshksoni
a7fe24a294 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/1145>
2023-09-20 13:54:48 +03:00
rajneeshksoni
4be24fdcaf 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/1145>
2023-09-20 13:54:48 +03:00
Sebastian Dröge
b12278e334 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/1326>
2023-09-16 10:59:27 +03:00
Seungha Yang
225482f7ed 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/1322>
2023-09-15 00:26:08 +09:00
Seungha Yang
1de7754616 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/1322>
2023-09-15 00:26:04 +09:00
Robert Ayrapetyan
18967dadbf gstwebrtc-api: drop guacamole
fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/417

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1317>
2023-09-11 19:21:41 +00:00
François Laignel
029fa9b8dc net/ndi: improve interoperability robustness
`quick-xml::reader::Reader::trim_text(true)` doesn't remove white spaces and
tabs from XML text. Besides, for interoperability robustness we also need to
remove carriage returns and line feeds.

Also improve the default capacities for the `SmallVec`s.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1321>
2023-09-11 06:22:41 +00:00
Mathieu Duponchelle
2381558169 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/1319>
2023-09-08 12:58:08 +00:00
François Laignel
9604dea90a net/ndi: add closed caption support
Closed caption support in NDI is described as a proposal in [1] & [2].

The proposal consists in encapsulating c608 or c708 closed caption in ADF
packets and pushing them in an XML tag as part of NDI Metadata.

This commit implements this proposal.

[1]: http://www.sienna-tv.com/ndi/ndiclosedcaptions.html
[2]: http://www.sienna-tv.com/ndi/ndiclosedcaptions608.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1320>
2023-09-07 14:28:24 +02:00
Robert Ayrapetyan
e83238b681 webrtcsink: fix TWCC extension adding
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1310>
2023-09-04 18:27:51 +00:00
Sebastian Dröge
b0b63e58f8 ndi: Comment out empty Opus handling and add FIXME comment
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
2023-08-29 12:21:38 +00:00
Sebastian Dröge
8d433761d1 Fix indentation of let-else blocks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
2023-08-29 12:21:38 +00:00
Taruntej Kanakamalla
de6d2e7f40 net/webrtc: rename whipwebrtcsink as whipclientsink
add a deprecation warning in whipsink to indicate it
should be used only for RTP content
add documentation in whipsink code regarding usage and
deprecation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
2023-08-26 10:53:30 +05:30
Sebastian Dröge
905da44958 Update to AWS SDK 0.30
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1313>
2023-08-25 09:46:52 +03:00
Andoni Morales Alastruey
3c1f05cdc3 webrtcsrc: document how to use the element for remote control
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
2023-08-10 17:43:51 +00:00
Andoni Morales Alastruey
3000b08ec7 webrtcsrc: add support for navigation events
This provides support GstNavigation events handling in webrtcsrc so that
a GStreamer client can be used to control remotely a GStreamer server,
similar to how the web client is capable of controlling a wpesrc.
This is part of a larger set of patches that require more work on the
sinks and sources.
server: d3d11screencapturesrc ! webrtcsink enable-data-channel-navigation=true
client: webrtcsrc enable-data-channel-navigation=true ! d3d11videosink

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
2023-08-10 17:43:51 +00:00
Loïc Le Page
e5e3dc6e19 net/webrtc/signaller: add property to get the connection client ID
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
2023-08-10 17:30:21 +02:00
Loïc Le Page
7af2ff0843 net/webrtc/signaller: advertise running producers in Listener mode
When starting a webrtcsrc-signaller client in Listener mode, only the producers
started after the client connection were advertised. All currently
running producers were ignored unlike the gstwebrtc-api behavior. This
commit now lists all running producers when the client Listener connects
and advertises them through the "producer-added" signal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
2023-08-10 17:30:21 +02:00
Sebastian Dröge
d688aeb184 Update versions to 0.12.0-alpha.1 2023-08-10 17:21:11 +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
Sebastian Dröge
5ee46a214c webrtc: Use #[repr(C)] to get a C-compatible layout for the Signaller struct
This is required by GObject for class/interface and instance structs and
the reason why implementing the `glib::ObjectInterface` trait is unsafe.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/397

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1297>
2023-08-09 10:32:44 +03:00
Sebastian Dröge
cac791a6ca aws/webrtc: Update to AWS SDK 0.56/0.29
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1295>
2023-08-07 20:03:51 +03:00
Sebastian Dröge
2591feb72e Update a couple of dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1294>
2023-08-07 11:42:32 +03:00
Sanchayan Maity
5b60ecbb18 net: webrtc/webrtchttp: Fix canceller usage
Commit 08b6251a added the check to ensure only one canceller at a time for net/webrtc.

In `whipsink` and since `whipwebrtcsink` picked up the same implementation, there exists a
bug around the use of canceller. `whipsink` calls `wait_async` while passing the canceller
as an argument. The path `send_offer -> do_post -> parse_endpoint_response` results in the
canceller being replaced in each subsequent call to `wait_async`. Since `wait_async` call
does not ensure one canceller, with the async call the use of canceller/abort was subtly
broken. Similarly, for `whepsrc`.

We really don't need to use `wait_async` inside `do_post` for any `await` calls. If the
root future viz. `do_post` with `wait_async` is aborted, the child futures will be taken
care of.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1290>
2023-08-04 10:01:11 +05:30
Mathieu Duponchelle
9680805bdb webrtcsink: don't forget to setup encoders for discoveries
The "encoder-setup" signal must also be emitted for the encoders
used in discovery pipelines in order for the default settings to
be applied.

This otherwise meant that for instance the x264 encoder would
use a 60 frames latency, greatly delaying startup.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1289>
2023-08-01 00:28:52 +02:00
Mathieu Duponchelle
dbeb65da06 webrtc/utils: fix typos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1289>
2023-08-01 00:28:32 +02:00
Sebastian Dröge
d4b3827efa webrtcsink: NVIDIA V4L2 encoders always require NVMM memory
And if the input is not like that then a corresponding converter must be
inserted.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1283>
2023-07-24 10:14:59 +00:00
Sebastian Dröge
31b1cb8ca6 Update minimum supported Rust version to 1.70
gtk-rs will update soonish too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1280>
2023-07-19 09:19:34 +03:00
Mathieu Duponchelle
9707bb89e6 webrtcsink: fix pipeline when input caps contain max-framerate
GstVideoInfo uses max-framerate to compute its fps, but this leads
to issues in videorate when framerate is actually 0/1.

Fix this by stripping away max-framerate from input caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1276>
2023-07-13 22:18:08 +02:00
Sebastian Dröge
0331522128 webrtcsink: Configure only 4 threads for x264enc
More threads can cause more slices to be created, and Chrome simply falls
apart if there are more than a few slices and fails decoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1275>
2023-07-13 16:59:43 +03:00