Commit graph

982 commits

Author SHA1 Message Date
Jordan Yelloz
96037fbcc5 webrtcsink: Updated livekitwebrtcsink for new signaller constructor
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:58 -07:00
Jordan Yelloz
730b3459f1 livekit_signaller: Added dual-role support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:49 -07:00
Guillaume Desmottes
60bb72ddc3 webrtc: janus: add joined-id property to the signaller
Fix #504

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 15:05:11 +01:00
Guillaume Desmottes
eabf31e6d0 webrtc: janus: rename RoomId to JanusId
Those weird ids are used in multiple places, not only for the room id,
so best to have a more generic name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 15:05:11 +01:00
Guillaume Desmottes
550018c917 webrtc: janus: room id not optional in 'joined' message
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 14:16:46 +01:00
Guillaume Desmottes
0829898d73 webrtc: janus: remove 'audio' and 'video' from publish messages
Those are deprecated and no longer used.

See https://janus.conf.meetecho.com/docs/videoroom and
https://github.com/meetecho/janus-gateway/blob/master/src/plugins/janus_videoroom.c#L9894

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 13:39:04 +01:00
Guillaume Desmottes
ec17c58dee webrtc: janus: numerical room ids are u64
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1478>
2024-02-28 11:56:44 +01:00
Yorick Smilda
563eff1193 Implement GstWebRTCAPI as class instead of global instance
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1373>
2024-02-27 12:30:13 +00:00
Jordan Yelloz
594400a7f5 webrtcsrc: Made producer-peer-id optional
It may be necessary for some signalling clients but the source element
doesn't need to depend on it.

Also, the value will fall back to the pad's MSID for the first argument
to the request-encoded-filter gobject signal when it isn't available
from the signalling client.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1477>
2024-02-26 13:41:40 -07:00
Xavier Claessens
f7ffa13543 janusvr: Add string-ids property
It forces usage of strings even if it can be parsed into an integer.
This allows joining room `"133"` in a server configured with string
room ids.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1466>
2024-02-26 11:10:00 +00:00
Xavier Claessens
23955d2dbb janusvr: Room IDs can be strings
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1466>
2024-02-26 11:10:00 +00:00
Sebastian Dröge
f563f8334b rtp: Add PCMU/PCMA RTP payloader / depayloader elements
These come with new generic RTP payloader, RTP raw-ish audio payloader
and RTP depayloader base classes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2024-02-23 14:43:45 +02:00
Maksym Khomenko
da21dc853d webrtcsink: extensions: separate API and signal checks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1469>
2024-02-20 19:29:46 +02:00
Maksym Khomenko
2228f882d8 webrtcsink: apply rustfmt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1469>
2024-02-20 19:29:28 +02:00
Xavier Claessens
2572afbf15 janusvr: Add secret-key property
Every API calls have an optional "apisecret" argument.

Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1465>
2024-02-16 14:04:59 +00:00
Sebastian Dröge
8ef12a72e8 rtpgccbwe: Don't reset PTS/DTS to None
The element is usually placed before `rtpsession`, and `rtpsession`
needs the PTS/DTS for correctly determining the running time. The
running time is then used to produce correct RTCP SR, and to potentially
update an NTP-64 RTP header extension if existing on the packets.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/496

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1462>
2024-02-14 08:05:54 +00:00
Jordan Yelloz
67b7cf9764 webrtcsink: Added sinkpad with "msid" property
This forwards to the webrtcbin sinkpad's msid when specified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-02-12 15:04:44 +00:00
Sebastian Dröge
b2d5ee48cd Update to async-tungstenite 0.25
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1455>
2024-02-11 11:31:24 +02:00
Sebastian Dröge
92891a61e8 Fix a couple of compiler/clippy warnings with --no-default-features
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1452>
2024-02-08 13:02:55 +02:00
Nirbheek Chauhan
cf5e7f6ed3 rtspsrc2: Add some top-level documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
7a1cd675c2 rtspsrc2: Fix RTCP send/recv in the multicast case
Don't use connect(), since that is incompatible with multicast.
Instead, drop received packets that are from senders we do not want.

Also set multicast loopback = false so we don't receive RTCP RRs from
ourselves and interpret them as RTCP SRs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
e59f3bbe58 rtspsrc2: Increase RTP timeout to 5 seconds, matching rtspsrc
Also fix some logging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
3e963e9239 rtspsrc2: Implement NetAddressMeta support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
42425abb69 rtspsrc: Factor out SDP → Caps, parse more attributes
This could be a struct of some kind derived from sdp_types::Media etc,
but this is fine for now.

Adds parsing of framesize, and fallbacks for missing or incomplete
rtpmap.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
437326ebfd rtspsrc2: Allocate a buffer pool for UDP RTP data
Control the size with a receive-mtu property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
44e49a06a0 rtspsrc2: Emit EOS if any ssrc gets a BYE packet or times out
This allows us to exit when the live-stream ends.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
975556c06b rtspsrc2: Allow a SETUP response without a Transports header
If we only send a single Transport in the Transports header, then the
server is allowed to omit it in the response. This has some strange
consequences for UDP transport: specifically, we have no idea what
addr/port we will get the packets from.

In those cases, we connect() on the socket when we receive the first
packet, so we can send RTCP RRs, and also so we can ensure that we
ignore data from other addresses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
086ffd7aff New RTSP source plugin with live streaming support
GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI]

Features:
* Live streaming N audio and N video
  - With RTCP-based A/V sync
* Lower transports: TCP, UDP, UDP-Multicast
* RTP, RTCP SR, RTCP RR
* OPTIONS DESCRIBE SETUP PLAY TEARDOWN
* Custom UDP socket management, does not use udpsrc/udpsink
* Supports both rtpbin and the rtpbin2 rust rewrite
  - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs)
* Properties:
  - protocols selection and priority (NEW!)
  - location supports rtsp[ut]://
  - port-start instead of port-range

Co-Authored-by: Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:18 +05:30
Bilal Elmoussaoui
0615a16124 Use workspace features for crates metadata/deps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1446>
2024-02-05 15:34:31 +01:00
Sebastian Dröge
91abc62ad0 webrtcsink: Fix new clippy warning
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:53:20 +02:00
Sebastian Dröge
1a55c70114 Switch git dependencies to explicitly name branch
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:51:36 +02:00
Sebastian Dröge
ffa830ae9b Update for GLib prelude re-organization
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1444>
2024-02-03 12:30:15 +02:00
Jordan Yelloz
311fda649f livekit_signaller: Added high-quality layer for video streams
This change addresses a cosmetic issue with livekit, where the
connection quality indicator seen by other users shows bad quality
unless the track is added with a high quality layer. The details of the
layer submitted aren't significant for this purpose.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1443>
2024-02-02 20:57:17 +00:00
Robert Ayrapetyan
916a8b959e doc: add http headers for webrtcsink signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
972b9e5474 doc: add docstrings for signaller object
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
7a72b2fc25 webrtcsink-signalling: add headers support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
François Laignel
91bfd0f7c3 webrtc: signallers: attempt to close the ws when an error occurs
This commit discards the early error returns in the send tasks to log the error
and attempt to close the websocket.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
François Laignel
f54d714afd webrtc: only use close() to close websockets
In the signaller clients and servers, the following sequence is used to close
the websocket (in the [send task]):

```rust
    ws_sink.send(WsMessage::Close(None)).await?;
    ws_sink.close().await?;
```

tungstenite's [`WebSocket::close()` doc] states:

> Calling this function is the same as calling `write(Message::Close(..))``

So we might think they are redundant and either could be used for this purpose
(`send()` calls `write()`, then `flush()`).

The result is actually is bit different as `write()` starts by checking the
state of the connection and [returns `SendAfterClosing`] if the socket is no
longer active, which is the case when a closing request has been received from
the peer via a [call to `do_close()`]). Note that `do_close()` also enqueues a
`Close` frame.

This behaviour is visible from the server's logs:

```
1. tungstenite::protocol: Received close frame: None
2. tungstenite::protocol: Replying to close with Frame { header: FrameHeader { .., opcode: Control(Close), .. }, payload: [] }
3. gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
4. gst_plugin_webrtc_signalling::server: connection closed: None this_id=cb13892f-b4d5-4d59-95e2-b3873a7bd319
5. remove_peer{peer_id="cb13892f-b4d5-4d59-95e2-b3873a7bd319"}: gst_plugin_webrtc_signalling::server: close time.busy=285µs time.idle=55.5µs
6. async_tungstenite: websocket start_send error: WebSocket protocol error: Sending after closing is not allowed
```

1: The server's websocket receives the peer's `Close(None)`.
2: `do_close()` enqueues a `Close` frame.
3: The incoming `Close(None)` is handled by the server.
4 & 5: perform session closing.
6: `ws_sink.send(WsMessage::Close(None))` attempts to `write()` while the ws
   is no longer active. The error causes an early return, which means that
   the enqueued `Close` frame is not flushed.

Depending on the peer's shutdown sequence, this can result in the following
error, which can bubble up as a `Message` on the application's bus:

```
ERROR: from element /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0: GStreamer encountered a general stream error.
Additional debug info:
net/webrtc/src/webrtcsrc/imp.rs(625): gstrswebrtc::webrtcsrc:👿:BaseWebRTCSrc::connect_signaller::{{closure}}::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0:
Signalling error: Error receiving: WebSocket protocol error: Connection reset without closing handshake
```

On the other hand, [`close()` ensures the ws is active] before attempting to
write a `Close` frame. If it's not, it only flushes the stream.

Thus, when we want to be able to close the websocket and/or to honor the closing
handshake in response to the peer `Close` message, the `ws_sink.close()`
variant is preferable.

This can be verified in the resulting server's logs:

```
tungstenite::protocol: Received close frame: None
tungstenite::protocol: Replying to close with Frame { header: FrameHeader { is_final: true, rsv1: false, rsv2: false, rsv3: false, opcode: Control(Close), mask: None}, payload: [] }
gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
gst_plugin_webrtc_signalling::server: connection closed: None this_id=192ed7ff-3b9d-45c5-be66-872cbe67d190
remove_peer{peer_id="192ed7ff-3b9d-45c5-be66-872cbe67d190"}: gst_plugin_webrtc_signalling::server: close time.busy=22.7µs time.idle=37.4µs
tungstenite::protocol: Sending pong/close
```

We now get the notification `Sending pong/close` (the closing handshake) instead
of `websocket start_send error` from step 6 with previous variant.

The `Connection reset without closing handshake` was not observed after this
change.

[send task]: 63b568f4a0/net/webrtc/signalling/src/server/mod.rs (L165)
[`WebSocket::close()` doc]: https://docs.rs/tungstenite/0.21.0/tungstenite/protocol/struct.WebSocket.html#method.close
[returns `SendAfterClosing`]: 85463b264e/src/protocol/mod.rs (L437)
[call to `do_close()`]: 85463b264e/src/protocol/mod.rs (L601)
[`close()` ensures the ws is active]: 85463b264e/src/protocol/mod.rs (L531)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
Taruntej Kanakamalla
50e905fe4b webrtc: conditional compile for features with 1_22 dependency
Few features being used in webrtcsink like
the signal `request-aux-sender` are introduced
to webrtcbin in gstreamer release 1.22.

Rename the feature gst1_22 to v1_22 for uniformity.

Add v1_22 to default features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1241>
2024-02-01 15:08:11 +05:30
Sebastian Dröge
f2a7a34abf rtp: gcc: Use x += ... instead of x = x + ... 2024-01-31 18:46:55 +02:00
Sebastian Dröge
4ad101b53b Use once_cell crate directly again
The glib crate does not depend on it anymore and also does not re-export
it anymore.

Also switch some usages of OnceCell to OnceLock from std.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
451d928026 webrtc: Update AWS signaller to http 1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sanchayan Maity
95c007953c webrtchttp: Allow audio or video caps to be specified as None with WHEP
We were setting audio and video caps by default even when the user
might have requested only video or audio. This would then result
in a `Could not reuse transceiver` error from the webrtcbin.

Fix this by allowing the user to specify audio or video caps as
None. This allows us to maintain the earlier behaviour for backward
compatibility while allowing the user to not request audio or video
as need be.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1433>
2024-01-18 15:43:19 +05:30
Sebastian Dröge
764143d971 webrtc: Remove unnecessary manual Send+Sync implementations for signallers
These are automatically implemented.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/483

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:25 +02:00
Sebastian Dröge
1af18f3028 webrtc: Require Send+Sync for signaller implementations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:01 +02:00
Eva Pace
80b58f3b45 net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller
The JanusVRWebRTCSink is a new plugin that integrates with the Video
Room plugin of the Janus Gateway, which simplifies WebRTC communication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2024-01-17 20:33:57 +00:00
Maksym Khomenko
773ebc7854 webrtcsrc: don't restrict RTP extensions to TWCC only
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
2024-01-17 07:34:01 +00:00
Sebastian Dröge
dfa95d8ed3 webrtc: Update to livekit-api / livekit-protocol 0.3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16 07:52:48 +00:00
Maksym Khomenko
fecbe01e06 webrtcsink: make 'extensions' property usage conditional
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
2024-01-16 07:13:56 +00:00
Sebastian Dröge
73a53e38c4 aws: s3: Disable remaining tests too for now
They fail state changes, which cases `GstHarness` to abort.
2024-01-16 09:13:41 +02:00
Arun Raghavan
fd3675aac0 aws: s3: Temporarily disable putobject tests
Disabling while we figure out why it's failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1429>
2024-01-15 21:43:25 -05:00
Arun Raghavan
8b18ca15b5 Revert "aws: Disable putobjectsink tests for now"
This reverts commit b128d127c2.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11 15:38:36 -05:00
Arun Raghavan
06213714c5 aws: putobjectsink: Fix a couple of minor log typos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11 15:38:36 -05:00
Nirbheek Chauhan
2d85048925 webrtc/signalling: We get the address when accepting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Nirbheek Chauhan
63b568f4a0 webrtc/signalling: Fix potential hang and FD leak
If a peer connects via TCP and never initiates TLS, then the server
will get stuck in the accept loop. Spawn a task when accepting a TLS
connection, and timeout if it doesn't complete in 5 seconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Maksym Khomenko
17f0b61576 webrtcsink: add payloader-setup signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-12-23 08:02:08 +00:00
Sebastian Dröge
b128d127c2 aws: Disable putobjectsink tests for now
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-22 13:25:12 +02:00
Arun Raghavan
6d47045a60 aws: s3sink: Fix spelling of debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Arun Raghavan
410d104ad6 aws: s3putobjectsink: Add a flush-on-error property
Makes sure we can send out data even if the pipeline shutdown in error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Arun Raghavan
12dbf50ddc aws: s3putobjectsink: Add some thresholds for flushing
Lets us connect when we perform a flush

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Arun Raghavan
a54b2dd39e aws: s3: Add a new awss3putobjectsink
When streaming small amounts of data, using awss3sink might not be a
good idea, as we need to accumulate at least 5 MB of data for a
multipart upload (or we flush on EOS).

The alternative, while inefficient, is to do a complete PutObject of
_all_ the data periodically so as to not lose data in case of a pipeline
failure. This element makes a start on this idea by doing a PutObject
for every buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 10:39:23 -05:00
Sebastian Dröge
81dd45c814 webrtc: Downgrade aws-smithy-http to 0.60
Version 0.61 was yanked from crates.io.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
2023-12-14 09:11:07 +02:00
Sebastian Dröge
2f2bf6ca8f webrtc: Update to aws-smithy-http 0.61
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:21:38 +02:00
Sebastian Dröge
0bae18fe0d rtp: Update to bitstream-io 2.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:17:51 +02:00
Sebastian Dröge
181bd13103 Update to async-tungstenite 0.24
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:17:11 +02:00
Guillaume Desmottes
6dfd1c1496 use new debug and parse API
Changes from https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1355

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1403>
2023-12-04 15:58:21 +01:00
Sebastian Dröge
f13574d8ed Update further AWS SDK crates to 1.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
2023-11-26 10:26:02 +02:00
Mathieu Duponchelle
cf1c7600a2 webrtcsink: don't panic on failure to request pad from webrtcbin
webrtcbin will refuse pad requests for all sorts of reasons, and should
be logging an error when doing so, simply post an error message and let
the application deal with it, the reason for the refusal should
hopefully be available in the logs to the user.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1399>
2023-11-24 19:53:38 +01:00
Sebastian Dröge
c3ced8c7e6 Update to AWS SDK 1.0 / 0.60 / 0.39
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1397>
2023-11-21 10:32:59 +02:00
Sebastian Dröge
1d9c89e3fe Update to AWS SDK 0.101 / 0.59 / 0.38 2023-11-20 10:13:13 +02:00
Sebastian Dröge
66c62d69b9 aws: Stop using deprecated aws_config function in the test 2023-11-18 10:16:24 +02:00
Taruntej Kanakamalla
43ee6bfc1c net/webrtc: add whipserversrc
Implement new signaller WhipServerSignaller
 - an http server using 'warp'
 - handlers for the POST, OPTIONS, PATCH and DELETE
 - fixed path `/whip/endpoint` as the URI
 - fixed value 'whip-client' as the producer peer id
 - fixed resource url `/whip/resource/whip-client`

Derive whipserversrc element from BaseWebRTCSrc
 - implement constructed method for ObjectImpl to set
  non-default signaller, i.e., WhipServerSignaller
 - bind the properties stun-server and turn-servers to those on
   the Signaller

Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller
 - it will be emitted by the webrtcsrc when the webrtcbin element is ready
 - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state
   and perform send with the answer sdp via the channel
 - the WhipServer will hold its HTTP response in POST handler until this signal
   is received or timeout which happens early

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
ed3aa740be net/webrtc: deprecate consumer-added on the signaller
add a new signal webrtcbin-ready in this place doing same
thing but can be used for both consumers and producers

Please note this change is only to the consumer-added
signal on the signaller interface.
The consumer-added signal on the webrtcsink is unchanged

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
2d3d03b4d3 net/webrtc: rename WhipSignaller as WhipClientSignaller
remove generalized names to accommodate for the WhipServer
- name the Signaller for whipsink as WhipClient
- name the Settings for whipsink as WhipClientSettings
- name the State for whipsink as WhipClientState

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
a0638ec983 net/webrtc: Extract BaseWebRTCSrc
Define a Base for all the webrtcsrc type elements
so they can all be derived from it. Similar to base
element defined for webrtcsink type elements

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Sebastian Dröge
dee27e35b7 Update to latest AWS SDK
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1395>
2023-11-17 11:22:29 +02:00
Sebastian Dröge
58723f2a8c Update to AWS SDK 0.36
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1394>
2023-11-15 17:20:58 +02:00
François Laignel
9250c592a7 ndi: don't accumulate meta with audio only streams
Currently, only closed caption metadata are supported. When the next video
frame is received, pending meta are dequeued and parsed. If close captions
are found, they are attached to the video frame.

For audio only streams, it doesn't make sense to enqueue metadata. They would
accumulate in `pending_metadata` and would never be dequeued.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/460

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1392>
2023-11-13 19:26:23 +01:00
Sebastian Dröge
39155ef81c ndisrc: Implement zerocopy handling for the received frames if possible
Also move processing from the capture thread to the streaming thread.
The NDI SDK can cause frame drops if not reading fast enough from it.

All frame processing is now handled inside the ndisrcdemux.

Also use a buffer pool for video if copying is necessary.

Additionally, make sure to use different stream ids in the stream-start
event for the audio and video pad.

This plugin now requires GStreamer 1.16 or newer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 13:22:48 +02:00
Sebastian Dröge
2afffb39dd ndi: Don't mark private type as public
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Sebastian Dröge
99d7cce0d6 ndi: Refactor frame structs to have static lifetimes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Sebastian Dröge
eb137ec6dc ndi: Remove wrong Clone impl on RecvInstance
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Arun Raghavan
771741c10c Revert "s3: tests: Remove emoji-based tests for now"
This reverts commit a49a5dcb11.

Now that hotdoc should work with emoji, let's bring the tests back.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1386>
2023-11-09 11:50:53 -05:00
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
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
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
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
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
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
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
Lieven Paulissen
05aa9fa431 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/1357>
2023-10-12 09:12:02 +03:00
François Laignel
022afa6375 ndi: use v210 encoding for cc and attach to video frame
The NDI closed captions specifications [1] define a variation where metadata is
attached to the video frame. This requires the AFD buffer to be v210 encoded.
This commit applies this strategy.

Another difference with previous version is that when an error occurs while
encoding or decoding a meta, next meta are also tried instead of failing
immediately.

Receiving closed captions as a standalone metadata is kept for interoperability
purposes. In this case, metadata is also expected to be v210 encoded.

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

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1356>
2023-10-11 21:25:29 +02:00
Maksym Khomenko
5b03f7d7b0 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/1355>
2023-10-11 11:54:51 +03:00
Sebastian Dröge
3fc6220009 Update to AWS SDK 0.33
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1354>
2023-10-09 11:28:05 +03:00
Taruntej Kanakamalla
245185d2f6 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/1351>
2023-10-06 13:11:46 +00:00
Maksym Khomenko
e4096b5157 webrtcsink: README: add documentation for custom signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1340>
2023-10-06 12:58:04 +03:00
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
Sebastian Dröge
ca51cf2509 webrtcsink: Translate force-keyunit events to force-IDR action signal for NVIDIA encoders
NVIDIA's v4l2 encoder elements don't handle the force-keyunit events but
instead provide a custom action signal based API for requesting a
keyframe.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1274>
2023-07-12 10:09:32 +00:00
Sebastian Dröge
bbd3d9ffe0 Remove unnecessary mut everywhere
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1273>
2023-07-11 10:09:35 +03:00
Sebastian Dröge
ee4aca3010 webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency on rtph26[45]pay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
2023-07-10 19:48:37 +03:00
Sebastian Dröge
957a28f239 webrtcsink: Set VP8/VP9 payloader based on payloader element factory name
Instead of checking the encoder's name. There are more VP8/VP9 encoders
than the ones from the vpx plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
2023-07-10 19:45:17 +03:00
Mathieu Duponchelle
1dd13c4812 webrtcsink: fix session_id / peer_id confusion
In a few places, for instance parameter names, peer_id was still used
when session_id was actually getting passed.

Go through all instances of peer_id in webrtcsink/imp.rs and address
those mix-ups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1269>
2023-07-07 05:33:30 +00:00
Bilal Elmoussaoui
dd2d7d9215 Use re-exported once_cell
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
2023-07-06 17:50:49 +03:00
Bilal Elmoussaoui
2cc98bf410 Adapt to glib::Continue rename
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
2023-07-06 17:50:49 +03:00
Sebastian Dröge
58adebb325 Fix a couple of typos 2023-07-06 13:50:17 +03:00
Olivier Crête
08b6251a7a webrtc-utils: Ensure there is only one cancellable call at a time
Since we only have one canceller at a time, panic if one try to
use it twice in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-07-05 21:43:17 +00:00
Olivier Crête
817b60a758 webrtc: Value.get() is already type checks in the property calls
GObject will have ensured we get a GValue of the right type.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-07-05 21:43:17 +00:00
Olivier Crête
793ee66afa webrtcsink: Add LiveKit WebRTC sink and signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-07-05 21:43:17 +00:00
Seungha Yang
1f0ce101eb awstranscriber: Tone down log message
It's not an ERROR case at all

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1263>
2023-06-28 23:57:54 +09:00
Sebastian Dröge
c350f3c2af webrtcink: Use correct property types for nvvideoconvert
These are enums and not plain integers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1256>
2023-06-26 14:48:58 +00:00
Mathieu Duponchelle
84a33ca7b9 webrtcsink: bring in signalling code from whipsink as a signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-06-16 00:32:56 +02:00
Mathieu Duponchelle
f00a169081 webrtcsrc: add twcc extension to codec-preferences when present
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1245>
2023-06-15 20:41:53 +00:00
Mathieu Duponchelle
1200ae0ee6 webrtcsink: improve debug
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
2023-06-14 22:27:15 +02:00
Mathieu Duponchelle
64056c5527 net/webrtc: improve documentation layout
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
2023-06-14 22:27:15 +02:00
Sebastian Dröge
8a7a1f519c webrtc: Update to fastrand 2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1240>
2023-06-09 09:36:51 +03:00
Mathieu Duponchelle
81ae675f2d webrtcsink: don't try to use cudaconvert if not present
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
2023-06-08 15:32:49 +02:00
Mathieu Duponchelle
7f78a8428e webrtcsink: dump discovery pipelines on state changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
2023-06-08 15:32:49 +02:00
Mathieu Duponchelle
7447d95f1b webrtc/signalling: fix race condition in message ordering
Spawning one task per message to send out instead of sending them out
sequentially from the one task used to poll the handler sometimes
resulted in peers receiving ICE candidates before SDP offers, triggering
hard to understand errors in the browser.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
2023-06-08 13:24:45 +02:00
Mathieu Duponchelle
de0f7a08fe gstwebrtc-api: fix firefox errors about more than two stun servers
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
2023-06-08 13:24:45 +02:00
Mathieu Duponchelle
cd4b90fef4 webrtcsink/utils: remove unused decoders field in DecodingInfo
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
2023-06-08 01:54:13 +02:00
Mathieu Duponchelle
271b583876 webrtcsink: avoid panic on unprepare from an async tokio context
.. and log an error with advice on how to dispose of elements properly
from a tokio runtime.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1218>
2023-06-07 19:57:19 +00:00
Sebastian Dröge
c65b3429ad Use MPL as license specifier for plugins only requiring GStreamer < 1.20
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1235>
2023-06-07 19:13:55 +03:00
Mathieu Duponchelle
fda5aed89f webrtcsink: encoded streams: address last review comments
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 16:05:28 +02:00
Thibault Saunier
ab1ec12698 webrtcsink: Add support for pre encoded streams
This is a first step where we try to replicate encoding conditions from
the input stream into the discovery pipeline. A second patch will
implement using input buffers in the discovery pipelines.

This moves discovery to using input buffers directly. Instead of trying
to replicate buffers that `webrtcsink` is getting as input with testsrc,
directly run discovery based on the real buffers. This way we are sure
we work with the exact right stream type and we don't need encoders to
support encoding streams inputs.

We use the same logic for both encoded and raw input to avoid having
several code paths and makes it all more correct in any case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:32:40 +02:00
Thibault Saunier
059cdecf7d webrtc: Unify the Codec structure between sink and source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:31:45 +02:00
Thibault Saunier
cf32d9d668 webrtc: Move make_element to the utils
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:31:45 +02:00
Thibault Saunier
ce42723ad2 webrtc: Minor documentation enhancement
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 15:31:45 +02:00
Mathieu Duponchelle
6346d5608e net/aws/transcriber: track discont offset in input stream
and add it up to subsequent transcripts.

This ensures synchronization is maintained even after the input stream
experiences a discontinuity and a gap in its timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1230>
2023-06-02 08:55:11 +00:00
Mathieu Duponchelle
80582923bb aws_kvs_signaller: don't force us-east-1 region
Instead use default region provider, with a fallback to us-east-1

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1228>
2023-05-30 16:04:27 +00:00
Edward Hervey
31b06e52ea rtpgccbwe: Improve packet handling
Both the delay-based *and* loss-based estimates should be computed instead of
just one. This ensures faster adaptation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1179>
2023-05-29 08:20:36 +00:00
François Laignel
4cc2498c24 webrtcsink: use spawn_blocking instead of call_async
In `webrtcsink`, we terminate a session by setting the session's pipeline to
`Null` like this:

```rust
    pipeline.call_async(|pipeline| {
        [...]
        pipeline.set_state(gst::State::Null);
        [...]
        // the following cvar is awaited in unprepare()
        cvar.notify_one();
    });
```

However, `pipeline.call_async` keeps a ref on the pipeline until it's done,
which means the `cvar` is notified before `pipeline` is actually 'disposed',
which happens in a different thread than `unprepare`'s. [`gst_rtp_bin_dispose`]
releases some resources when the pipeline is unrefed. In some cases, those
resources are actually released after the main thread has returned, leading
various issues.

This commit uses tokio runtime's `spawn_blocking` instead, which allows owning
and disposing of the pipeline before the `cvar` is notified.

[`gst_rtp_bin_dispose`]: https://gitlab.freedesktop.org/gstreamer/gstreamer/-/blob/main/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpbin.c#L3108

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1225>
2023-05-26 14:23:03 +02:00
Mathieu Duponchelle
a20855dfd9 webrtcsink: expose consumer-pipeline-created signal
This signal is emitted as soon as the pipeline for each consumer
is created, and can be used by applications that require a greater
level of control over webrtcsink's internals.

An example is also provided to demonstrate usage

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1220>
2023-05-25 13:15:52 +02:00
Sebastian Dröge
a27be7d054 net: Update to AWS SDK 0.28
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1224>
2023-05-25 13:23:49 +03:00
François Laignel
e62e9f5bd4 webrtcsink: adapt commit "abort stats collection before stopping the Signaller"
Adapt a commit [1] that was introduced as part of the forward port of the MR
'add signal "request-encoded-filter"' [2].

The deadlock said commit was fixing doesn't happen on main branch due to
changes in the element design: the Sessions are no longer aborted with the
element `State` held. However, we want to ensure the stats collection task
is terminated when the `webrtcbin` element returns from the Ready to Null
transition, meaning that the related resources are released.

[1]: gstreamer/gst-plugins-rs!1176 (0e6b9df9)
[2]: gstreamer/gst-plugins-rs!1176

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1222>
2023-05-24 21:35:39 +02:00
Sebastian Dröge
e3c46b40a0 whipsink: Request pads with webrtcbin's pad templates and not our own
It's invalid to request pads with a pad template that is not part of the
element, and results in a critical warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1223>
2023-05-24 14:14:32 +00:00
Mathieu Duponchelle
44a395f134 webrtcsink: further refactor connection to stats signals
- Stop passing webrtcbin around without using it

- Stop using glib::closure as clippy complains when using a unit type
  default-return

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1217>
2023-05-24 13:35:26 +02:00
Mathieu Duponchelle
e13124a426 webrtcsink: fix stats_sigid logic
First off, we just created the session, we know stats_sigid is None
at this point.

Second, don't first assign the result of connecting on-new-ssrc to the
field, then the result of connection twcc-stats, that simply doesn't
make sense.

Finally, actually check that stats_sigid *is* None before connecting
twcc-stats, as I understand it this must have been the original
intention / behavior.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1217>
2023-05-24 13:35:26 +02:00
Mathieu Duponchelle
ccf076ed1e webrtcsink: don't panic in twcc-stats callback
If webrtcbin was disposed of at this point, simply return

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/345
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1217>
2023-05-24 13:35:26 +02:00
François Laignel
9a59763df1 webrtcsink: wait for Sessions to end
`State::finalize_session()` asynchronously sets the Session pipeline to Null.
In some cases, sessions `webrtcbin` could terminate their transition to Null
after `webrtcsink` had reached Null.

This commit adds a set of `finalizing_sessions`. When the finalization process
starts, the session is added to the set. After `webrtcbin` has reached the Null
state, the session is removed from the set and a condvar is notified.

In `unprepare`, `webrtcsink` loops until the `finalizing_sessions` set is
empty, awaiting for the condvar to be notified when it's not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1221>
2023-05-24 10:18:47 +02:00
François Laignel
b68e2a1ed0 webrtcsink: remove unneeded mut
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1221>
2023-05-24 10:18:43 +02:00
Thibault Saunier
04e35e86d6 webrtcsrc: Do not pass raw caps in the transceiver
That was not making sense.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1214>
2023-05-18 18:23:56 +03:00
Thibault Saunier
e73d7082a6 webrtcsrc: Fix caps used when creating transceiver
We used to pass all media keys and attributes to the caps which
incorrect. Instead we should be using only the keys from the map
and remove all information related to rtcp which is irrelevant
to create the transceiver.

This also simplifies the code.

New caps look like:

```
Caps(
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 96,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP8",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 102,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 104,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 106,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 108,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 127,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 39,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 98,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "0",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 100,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "2",
    },
)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1214>
2023-05-18 18:23:56 +03:00
François Laignel
7ba0073052 use Pad builders for optional name definition
Also, apply auto-naming in the following cases

* When building from a non wildcard-named template, the name of the template is
  automatically assigned to the Pad. User can override with a specific name by
  calling `name()` on the `PadBuilder`.
* When building with a target and no name was provided via the above, the
  GhostPad is named after the target.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/448
Auto-naming discussion: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1255#note_1891181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
2023-05-12 12:55:31 +02:00
François Laignel
8e93d294e5 Update to argumentless {Bin,Pipeline}::new
See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/449

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
2023-05-12 12:55:31 +02:00
François Laignel
680d5221db net/webrtc: src: add signal "request-encoded-filter"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 12:02:15 +02:00
François Laignel
092ae1fec8 net/webrtc: sink: add signal "request-encoded-filter"
The new "request-encoded-filter" signal is emitted when the encoder and related
elements are added to the pipeline. When defined, the element returned by the
signal is inserted between the encoder and the payloader.

The transformation can be reverted using the [insertable streams API] on the
receiver side.

[insertable streams API]: https://developer.mozilla.org/en-US/docs/Web/API/Insertable_Streams_for_MediaStreamTrack_API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 11:17:32 +02:00
François Laignel
dc5ddd3022 net/webrtc: sink: abort stats collection before stopping the Signaller
In some rare cases, the webrtc-test entered a deadlock while executing
`WebRTCSink::unprepare`. Attaching gdb to a blocked instance showed:

* `gstrswebrtc::signaller:👿:Signaller::stop()` parked, waiting for a
  `Condvar` in `Signaller::stop()`. This was most likely awaiting for the
  receive task to complete while it was locked in `element.end_session()`.
  This code path is triggered from `unprepare` with the `State` `Mutex` locked.
* `webrtcsink:👿:WebRtcSink::process_stats` waiting for a contended `Mutex`,
  which is also the `State` `Mutex`. This prevented completion of the signal
  `gst_webrtc_bin_get_stats`.

This commit aborts the task in charge of periodically collecting stats and
ensures any remaining iteration completes before requesting the Signaller to
stop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 10:26:11 +02:00
François Laignel
eca269cbf2 net/webrtc: src: don't set stun-server on webrtcbin when our property is None
... otherwise an error occurs about the stun-server address being an empty
string which doesn't comply with the expected address format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 10:26:07 +02:00
Sebastian Dröge
cb5b527d74 Update to AWS SDK 0.27 and async-tungstenite 0.22
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1199>
2023-05-02 15:30:00 +03:00
Sebastian Dröge
5451035215 Update async-tungstenite and AWS SDK dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1187>
2023-04-21 10:48:10 +00:00
Sebastian Dröge
cc3646640e Fix a couple of new Rust 1.69 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1186>
2023-04-20 16:47:45 +03:00
Edward Hervey
721d17e181 rtpgccbwe: Don't process empty lists
The structure parsing could result in an empty vector. Don't do any processing
since the loss code assumes it's non-empty for average estimates which would
result in weird/invalid results.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1181>
2023-04-15 19:35:27 +02:00
Mathieu Duponchelle
dbdb9bc164 webrtcsink: fix navigation data channel
At some point, presumably recently, the data channel stopped being
requested in Ready, making webrtcbin refuse to create it.

There was quite a lot of churn recently so I couldn't pinpoint the
breaking commit easily.

Fix by simply restoring the correct behavior of requesting the channel
after going to the Ready state

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/341

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1180>
2023-04-14 14:26:22 +02:00
Mathieu Duponchelle
f1fd8d84c3 webrtc: extract a BaseWebRTCSink
For documentation purposes, AwsKVSWebRTCSink should not inherit from
another element.

+ Mark base class as plugin API and update plugin cache

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:06:59 +00:00
Loïc Le Page
dba91bceca webrtc: fix documentation after signaller interface changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1175>
2023-04-12 20:19:22 +02:00
Thibault Saunier
8f2273328b webrtcsrc: Return bool en 'end-session' as required
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1172>
2023-04-12 12:17:56 +00:00
Sebastian Dröge
5dcdf645d6 net: ndi: Update to libloading 0.8
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1173>
2023-04-12 11:03:05 +03:00
Mathieu Duponchelle
f366c20869 awstranscriber: fix what we send over for translations
Prior to this commit, we were sending over words concatenated together
with no separators, for instance "Idon'twanttobeanemperor".

The translation service seems clever enough to translate the contents
anyway, but there is no reason to make its task harder than necessary,
and it didn't re-add separators when the target language was the same as
the source language, which resulted in less than ideal output.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
2023-04-10 20:47:12 +00:00
Mathieu Duponchelle
408fd2030c awstranscriber: slight debug improvement
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
2023-04-10 20:47:12 +00:00
Guillaume Desmottes
403004a85e fix typos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
2023-04-10 13:35:32 +02:00
Mathieu Duponchelle
a455819871 webrtcsink: fix tracking of signaller state
For the signaller to get stopped, we need to remember that we started it
in the first place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
3368f55a88 webrtcsink: don't return value from error closure
the signal doesn't expect a return value, which meant we were panicking
as soon as the signaller tried to report an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
58c8c0edc7 webrtc: signaller iface: fix session-ended vs end-session confusion
Session ending is bidirectional: the signaller can tell the sink that a
session was ended, and the sink can tell the signaller to end a session.

As such, two signals are needed, before this patch the second case was
not working as in essence the sink was telling itself that a session was
ended, and obviously failing to even find it when trying to end it again.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
2023-04-10 07:58:10 +03:00
Tim-Philipp Müller
7c30430320 webrtc-api: replace LICENSE file symlink with copy
As in !1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1169>
2023-04-08 17:22:37 +01:00
Matthew Waters
e69b4b7f45 webrtc/signaller/iface: give variables appropriate names
Rather than arg0, arg1, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-07 09:58:13 +10:00
Matthew Waters
4f4e5f0d75 webrtcsink/signaller: don't call signals while having state/settings locked
It is a recipe for deadlocks if the signal callback calls back into
webrtcsink in some way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-07 09:58:13 +10:00
Matthew Waters
1c61e46f37 webrtcsink: privatise signalling functions
The functionality is now access through the relevant signals instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-07 09:58:13 +10:00
Matthew Waters
2ac560975c webrtc/signaller: emit the relevant signals instead of the interface vtable
In order to support the use case of an external user providing their own
signalling mechanism, we want the signals to be used and only if nothing
is connected, fallback to the default handling.  Calling the interface
vtable directly will bypass the signal emission entirely.

Also ensure that the signals are defined properly for this case. i.e.
1. Signals the the application/external code is expected to emit are
   marked as an action signal.
2. Add accumulators to avoid calling the default class handler if
   another signal handler is connected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-07 09:58:13 +10:00
Matthew Waters
343b659755 webrtc/signaller: remove SignallableImplExt
This pattern is used for subclassing and calling parent class/interface functions.
However that is not useful for the signaller object.
1. The signals are the API contract and should instead be used by
   webrtcsrc/sink to ask or provide outside for/with information.
2. The default case (no signal attached)is instead handled by default class
   handlers that call directly using the relevant rust trait.  No parent
   (GObject) vfuncs necessary.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-07 09:58:13 +10:00
Matthew Waters
b6e78b5f04 webrtcsink: expose signaller as a property
in the process move the signaller field to the settings struct

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-07 09:58:13 +10:00
Thibault Saunier
8236f3e5e7 webrtcsink: Port to the 'webrtcsrc' signaller object/interface
With contributions from:
Matthew Waters <matthew@centricular.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
2023-04-07 09:03:47 +10:00
Seungha Yang
762fb86ce7 awstranscriber: Reset start_time per task
Otherwise wrong start time can be assigned if the element is
reused with state change

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1159>
2023-04-05 18:22:59 +00:00
Sebastian Dröge
9cb211470f ndisrc: Fix copying of raw video frames with different NDI/GStreamer strides
And also don't copy each line twice for single-plane formats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1158>
2023-04-05 16:45:48 +03:00
Loïc Le Page
f17622a1e1 webrtc: Add gstwebrtc-api subproject in net/webrtc plugin
This subproject adds a high-level web API compatible with GStreamer
webrtcsrc and webrtcsink elements and the corresponding signaling
server. It allows a perfect bidirectional communication between HTML5
WebRTC API and native GStreamer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/946>
2023-04-04 16:29:44 +02:00
Tim-Philipp Müller
8845f6a4c6 git: replace LICENSE file symlinks with copies
Git will de-duplicate the contents for us anyway, and
symlinks can cause problems with some versions of git
and also on Windows.

https://github.com/mesonbuild/meson/issues/11646
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4326

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1157>
2023-04-04 14:26:37 +01:00
Seungha Yang
4000d60305 awstranscriber: Avoid too large initial GAP event
Initialized GstSegment.position is always zero

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1154>
2023-04-03 13:05:15 +00:00
Mathieu Duponchelle
15e1844956 webrtcsink: fix calculation of fec_ratio with multiple encoders
In this context, the bitrate variable is for all encoders, but the
max_bitrate field is per encoder. To calculate a proper FEC ratio, we
need to scale max_bitrate to the number of encoders.

+ Also clamp the fec-percentage that we set on the transceiver for extra
  safety

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1151>
2023-03-31 12:19:07 +00:00
Sebastian Dröge
315e53f064 webrtc: Update to AWS SDK 0.55/0.25
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
2023-03-31 09:12:26 +00:00
Sebastian Dröge
6fe806c2b5 aws: Update to AWS SDK 0.55/0.25
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
2023-03-31 09:12:26 +00:00
David Revay
002a70a2a4 chore(webrtcsink): fix max-bitrate blurb and nick
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
2023-03-28 16:11:05 +11:00
Vivia Nikolaidou
7a1b2d97d4 webrtcsink: Add ice-transport-policy option
Can be used to force relay ICE candidates, ensuring TURN server is used.
Proxy to the corresponding setting in webrtcbin,

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1143>
2023-03-27 16:12:13 +03:00
François Laignel
2b32d00589 net/aws/transcriber: use two queues for sending transcript items
* A queue dedicated to transcript items not intended for translation.
* A queue dedicated to transcript items intended for translation. The items are
  enqueued after a separator is detected or translate-lookahead was reached.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16 20:29:31 +01:00
François Laignel
5a5ca76d9d net/aws/transcriber: desambiguify SrcPad output items queue
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16 12:41:07 +01:00
François Laignel
162db2f3b9 net/aws/transcriber: fix translate lookahead
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16 12:39:15 +01:00
François Laignel
d5d6a4daf9 net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad
... as translate-lookahead and TranslateSrcPad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16 12:37:31 +01:00
François Laignel
3b3f0c1a29 net/aws/transcriber: fix transcript-lookahead prop nick
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1136>
2023-03-14 21:11:33 +01:00
François Laignel
299e25ab3c net/aws/transcriber: translate: optional experimental translation tokenization
This commit adds an optional experimental translation tokenization feature.
It can be activated using the `translation_src_%u` pads property
`tokenization-method`. For the moment, the feature is deactivated by default.

The Translate ws accepts '<span></span>' tags in the input and adds matching
tags in the output. When an 'id' is also provided as an attribute of the
'span', the matching output tag also uses this 'id'.

In the context of close captions, the 'id's are of little use. However, we can
take advantage of the spans in the output to identify translation chunks, which
more or less reflect the rythm of the input transcript.

This commit adds simples spans (no 'id') to the input Transcript Items and
parses the resulting spans in the translated output, assigning the timestamps
and durations sequentially from the input Transcript Items. Edge cases such as
absence of spans, nested spans were observed and are handled here. Similarly,
mismatches between the number of input and output items are taken care of by
some sort of reconcialiation.

Note that this is still experimental and requires further testings.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-14 13:48:32 +00:00
François Laignel
743e97738f net/aws/transcriber: add translation request src pads
This commit adds an optional transcript translation feature implemented as
request src Pads.

When requesting a src Pad, the user can specify the translation language code
using Pad properties 'language-code'.

The following properties are defined on the Element:

- 'transcribe-latency': formerly 'latency', defines the expected latency for
  the Transcribe webservice.
- 'translate-latency': defines the expected latency for the Translate
  webservice.
- 'transcript-lookahead': maximum transcript duration to send to translation
  when a transcript is hitting its deadline and no punctuation was found.

When the input and output languages are the same, only the 'transcribe-latency'
is used for the Pad. Otherwise, the resulting latency is the addition of
'transcribe-latency' and 'translate-latency'.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
2023-03-14 13:48:32 +00:00
Sebastian Dröge
4eccd30ce2 Revert "aws: Temporarily enable the default features of the test-with crate"
This reverts commit 42116b5bce.
2023-03-14 13:28:28 +02:00
Sebastian Dröge
42116b5bce aws: Temporarily enable the default features of the test-with crate
Version 0.9.4 fails compiling without them enabled.

See https://github.com/yanganto/test-with/pull/57
2023-03-14 09:19:26 +02:00
Sebastian Dröge
c1bac30694 webrtc: Update to aws 0.54/0.24
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1131>
2023-03-11 09:37:14 +02:00
Mathieu Duponchelle
584392049c net/webrtc: implement AWS KVS signaller
And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink.

This adds support in webrtcsink for processing a consumer offer, instead
of producing one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-09 15:39:09 +00:00
Sebastian Dröge
fc5ed15af5 Update for gst::Element::link_many() and related API generalization
Specifically, get rid of now unneeded `&`.
2023-03-09 16:46:52 +02:00
François Laignel
b9cd71d8eb net/aws/transcriber: fix eos not being sent
For eos to be sent from the srcpad task loop, we need to go through `dequeue`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1122>
2023-03-09 13:07:03 +01:00
François Laignel
2ea9f147ab net/aws/transcriber: fix deadlock when the pipeline is interrupted
... also makes sure to abort the taks_iter Future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1122>
2023-03-09 13:07:03 +01:00
Sebastian Dröge
3ef8a48ded Fix a few new clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1120>
2023-03-07 08:47:01 +00:00
Vivia Nikolaidou
cd74d01324 ndisinkcombiner: Properly handle caps changes
We are caching one video buffer, so previously we were changing the src
caps one buffer too early.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1110>
2023-03-01 12:30:54 +00:00
François Laignel
4a988aaeb8 net/aws/transcriber: use a TranscriberLoop struct
This helps gather together the details related to the `TranscriberLoop`.
One difference with previous implementation is that the ws `Client` is
build each time the loop is started instead of being reused. With the new
approach, we don't keep the connection open after EOS and we should be
more resistant in case of a connection failure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel
f1a080c94e net/aws/transcriber: own transcription items
So that we can avoid copying the content.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel
36ae29d746 net/aws: enqueue transcribed buffers within the ws loop
Instead of sending transcription events to the src pad loop, this commit
enqueues the transcribed buffers immediately in the ws loop, then notifies
the src pad loop. The src pad loop is only in charge of dequeuing the buffers.

This should help with upcoming evolutions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel
00153754bb net/aws: use aws-sdk-transcribestreaming
Switch from manual webservice client impl to `aws-sdk-transcribestreaming`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
François Laignel
57f365979c net/aws: remove aws_ from the aws_transcribe* folder names
Those folders reside under `aws`, so there's shouldn't be any confusion.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
Thibault Saunier
ce3bb2f1d4 Add a webrtcsrc element
Updating the docker image to include:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 20:50:15 -03:00
Thibault Saunier
0ae637f531 webrtcsink: Move RUNTIME to the crate so it can be reused
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 17:57:14 -03:00
Thibault Saunier
4ec441560b webrtc: Enhance debug messages when using unknown peer ID
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 19:28:51 +00:00
Matthew Waters
542c7e12b8 webrtcsink: also support nvvidconv in lieu of nvvideoconvert
nvvideoconvert may not exist and nvvidconv might on some Jetson
platforms.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1107>
2023-02-28 10:12:36 +11:00
Sebastian Dröge
9fc1404415 Update minimum supported Rust version to 1.66
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
2023-02-20 11:09:01 +02:00
Arun Raghavan
487d7fb26b hlssink3: Allow GIOStream signal handlers to return None
If creating a playlist or fragment stream fails (disk is full, the
directory is removed, ...), we will currently crash because the signal
handler expects a non-None GIOStream. The actual callback is allowed to
return None values and we handle this in the caller, so let's not have
this restriction on the signal handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1093>
2023-02-14 11:25:44 -05:00
Sebastian Dröge
04e101c605 Optimize various error message / debug message formatting
Directly make use of format strings instead of formatting a string
beforehand and then passing it to the macros.
2023-02-13 11:50:57 +02:00
Arun Raghavan
39e0acb55a hlssink3: Fix case on unspecified playlist type nick for consistency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1089>
2023-02-10 23:07:12 +00:00
Seungha Yang
6420fe43da rtpav1pay: Fix Leb128Bytes size parsing
There are multiple ways of encoding the value, and don't assume
that bitstream used the way used in this plugin. Instead, count
the number of used bytes.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/312
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1090>
2023-02-10 18:47:52 +00:00
Sebastian Dröge
ac8afc4ac0 Update to async-tungstenite 0.20
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1087>
2023-02-10 13:03:07 +02:00
Sebastian Dröge
1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +02:00
rajneeshksoni
994c79569e awss3sink: Add properties to set content-Type and content-disposition.
for uploaded object default content-type is set to binary/octet-stream,
which is correct.
metadata cannot be used to set content-type and content-disposition as
setting metadata add a prefix x-amz-meta to key
e.g. setting metadate "content-type=video/mp4" actually set value as
x-amz-meta-content-type. So these has to be seaprate property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1085>
2023-02-09 19:04:07 +00:00
rajneeshksoni
0f383a6545 hlssink3: Allow setting i-frame-only playlist.
HLS allows manifest where all segments are single ifames.
This manifest requires `EXT-X-I-FRAMES-ONLY` tag in the
manifest.
I-FRAMES-ONLY playlist segments are video only segments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1070>
2023-02-08 14:04:46 +00:00
Sebastian Dröge
0ed74d0aa4 rtpgccbwe: Don't use clamp() if there's no clear min/max value
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/305

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1078>
2023-02-06 21:56:46 +02:00
Sanchayan Maity
6006a0ba36 aws/s3hlssink: Fix deadlock on EOS
In state change to NULL, we take state lock and call stop. When stop
is called, we will try to upload queued segments in S3 request thread.
That tries to take the state lock again and deadlocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1076>
2023-02-03 19:09:18 +05:30
Sanchayan Maity
41aa1e51da aws/s3hlssink: Use factory name when checking name of child element
Commit ad3f1cf fixed the name of hlssink child element to be the same
for hlssink2 and hlssink3. However, we rely on element name to return
boolean in case of hlssink3 or None in case of hlssink2 as the return
value of the delete-fragment closure.

Fix this by using the factory name instead of the element name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1076>
2023-02-03 19:08:40 +05:30
Sebastian Dröge
5506f8001e rtpav1pay: Add support for tu/frame aligned input
In this case every buffer can be sent out immediately and makes up a
whole frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
194c4e9e9f rtpav1pay: Consider the marker flag to output packets immediately at the end of a frame
Otherwise it is necessary to wait for the beginning of the following
frame, which unnecessarily increases the latency.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/255

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
49350f738f rtpav1depay: Fix depayloading of packets starting with a leading OBU fragment followed by more OBUs
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/288

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
1756d7a516 rtpav1depay: Fix error handling
Don't error out immediately on errors anymore but try again with the
next packet.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/289

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
ed4e9a50d5 rtpav1depay: Set DISCONT flag on buffers following a corrupted packet
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
d6cb9d72d8 rtpav1depay: Don't output full TUs but just OBUs as they come
Simplifies state tracking and potentially reduces latency as it's not
necessary to wait until all fragments of an OBU are received.

The last OBU of a TU is marked with the marker flag to allow parsers to
detect this without first seeing the beginning of the next TU.

Also use a simple `Vec` for collecting complete OBUs instead of a
`gst_base::Adapter` as this reduces the number of allocations.

And also handle invalid packets a little bit more gracefully.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/244

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
560bdc4cb7 Update for glib API changes 2023-01-31 12:24:07 +02:00
Sebastian Dröge
a1cce9b796 aws: Update to AWS SDK 0.54/0.24
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1066>
2023-01-27 22:10:23 +02:00
Sebastian Dröge
3b4c48d9f5 Fix various new clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1062>
2023-01-25 10:31:19 +02:00
Arun Raghavan
ad3f1cf534 aws: s3hlssink: Fix the name of the hlssink child element
It's easier to set child element properties if the name doesn't depend
on the factory.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1061>
2023-01-24 18:56:46 +00:00
Sebastian Dröge
2c386fb792 Update for various deprecated APIs 2023-01-22 20:07:26 +02:00
Sebastian Dröge
4582ae91ab Move remaining plugins to ParamSpec builders
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1054>
2023-01-21 18:34:55 +02:00
Sebastian Dröge
458b2386ed Update for glib API changes 2023-01-21 18:13:48 +02:00
Sebastian Dröge
7cfd570c15 onvif: Update for allocation query caps API changes 2023-01-19 16:38:06 +02:00
Sebastian Dröge
812df78b75 webrtcbin: Update for StreamProducer API changes 2023-01-16 16:36:41 +02:00
Sebastian Dröge
6132788b02 Update for caps/structure-related string API changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1048>
2023-01-15 22:58:44 +02:00
Sebastian Dröge
0c954135a3 aws: Update to AWS SDK 0.53/0.23
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1047>
2023-01-14 18:58:30 +02:00
Mathieu Duponchelle
1a8abde884 webrtcsink: fix panic on pre-bwe request error
We dispose of consumer pipelines asynchronously, potentially after the
session objects have been disposed of.

As session objects are the owner of the cc element, it is entirely
possible for the bwe-request signal to get emitted after cc has been
disposed of, as the closure only takes a weak reference to it.

Fix by simply checking if cc is None

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1044>
2023-01-11 15:09:45 +00:00
Sebastian Dröge
be72fefb18 reqwest: Update for API changes 2023-01-06 12:52:30 +02:00
Sebastian Dröge
781fd1df9a aws: Update to test-with 0.9
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1035>
2023-01-05 12:35:42 +02:00
Sebastian Dröge
27435ad82e Update for API changes 2023-01-05 12:33:54 +02:00
rajneeshksoni
d846f527af awss3hlssink: Add stats property.
application can monitor the progress of hls segment generation
and upload progress.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1022>
2023-01-04 12:36:13 +00:00
Philippe Normand
0fd63ece7d rtpav1depay: Implement srcpad set_caps
Without this auto-pluggers such as decodebin or parsebin will be unable to
process AV1 RTP payloads.

Tested with: `videotestsrc num-buffers=50 ! videoconvert ! av1enc ! av1parse ! rtpav1pay ! queue ! decodebin3 ! videoconvert ! queue ! autovideosink`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1034>
2023-01-03 19:35:45 +02:00
Zhao, Gang
9fa838e366 webrtc: Fix rustfmt errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
2022-12-27 11:12:54 +02:00
Zhao, Gang
877a9bd7f3 webrtc: Share runtime between webrtcsink and signaller crates
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
2022-12-26 23:10:40 +00:00
Zhao, Gang
1ffeb4d44d webrtc: Move from async-std to tokio
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
2022-12-26 23:10:40 +00:00
Zhao, Gang
2bc29c1fd3 webrtc: examples: Update package-lock.json
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
2022-12-26 23:10:40 +00:00
Sebastian Dröge
4e444a066c aws: Update to AWS SDK 0.52/0.22
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1020>
2022-12-18 07:54:30 +00:00
Mathieu Duponchelle
e5360ff431 webrtc/README: update command to run the signalling server
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/277

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1012>
2022-12-13 12:47:26 +01:00
Sebastian Dröge
3f904553ea Fix various new clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1011>
2022-12-13 11:43:16 +02:00
Sebastian Dröge
289e8a08c3 webrtchttp: Remove unnecessary clippy warning override
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1009>
2022-12-12 14:32:12 +02:00
Sebastian Dröge
fb42cd8a0f net: Update to async-tungstenite 0.19
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1005>
2022-12-11 12:54:24 +02:00
Sebastian Dröge
9b964db4c9 whipsink: Handle offer creation errors more gracefully
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 12:15:55 +02:00
Sebastian Dröge
8452cd9efa webrtchttp: Fix missing import for docs build
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 12:10:53 +02:00
Sebastian Dröge
9c31344bbc webrtchttp: Don't use let-else for now
We still support Rust 1.63.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 12:08:57 +02:00
Sebastian Dröge
5dc52975ff webrtchttp: Fix formatting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 12:07:09 +02:00
Sanchayan Maity
40680a47ab webrtchttp: Use tokio runtime for spawning thread used for candidate offer
While at it, we had a bug in whepsrc where for redirect we were
incorrectly calling initial_post_request instead of do_post. Fix
that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 12:27:07 +05:30
Sanchayan Maity
d18761892e webrtchttp: Use a proper Rust type name for ICE transport policy
We don't need to namespace here but can just use the Rust namespaces.
Only the GType name has to stay like it is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
2eba3b321e webrtchttp: Do not import element_imp_error
element_imp_error and such macros should not be imported but rather
only be accessed via gst namespace.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
0b1b8b91b9 webrtchttp: Do not block webrtcbin signal handlers for sending candidates
While at it, drop the OPTIONS request in WHIP sink. This was not really
required. See section 4.4 of the spec
https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html#name-stun-turn-server-configurat

Also introduce a new error type and distinguish between a future being
aborted or returning an error.

We call abort only during shutdown and hence except for the DELETE
resource request being aborted, other waits on future should not
be fatal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Alba Mendez
db39370701 webrtchttp: whipsink: construct TURN URL correctly
Right now the code manually pieces together the components
in a String for efficiency. When credentials contain special
characters this can result in invalid URLs, so do it the proper
way (with Url::parse + format) to make sure components are escaped
as needed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
9fb058d5bc webrtchttp: Drop unused dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
b5daa92c9d webrtchttp: Implement timeout for waiting on futures
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
cc7419308b webrtchttp: whipsink: Add candidates when sending the offer
WHIP endpoint providers like Cloudflare do not support Trickle ICE
and need candidates to be send along with the initial offer. Instead
of sending the offer in create-offer promise, send it once the ICE
candidates have been gathered.

While at it add properties to set STUN and TURN server along with the
ICE transport policy as at least when testing the Cloudflare WHIP
endpoint seems unreachable without it. This has also been observed
with Cloudflare provided demos.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
b992596236 webrtchttp: whipsink: Miscellaneous clean up
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
b427cb6a3d webrtchttp: Factor out the common bits for WHIP and WHEP
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Sanchayan Maity
6be5796888 Add a WebRTC WHEP source element
This implements WHEP specification based on
https://datatracker.ietf.org/doc/html/draft-murillo-whep-00

and has been tested with Cloudflare.

Server offers are likely to be removed from the WHEP specification
in upcoming revisions, to avoid compatibility issues. None of the
commercial services implementing WHEP support server initiated offers.
So we only support client side initiated offers.

Follows session setup and tear down as covered in Figure 1, Section 3
of the specification.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
2022-12-05 11:04:45 +05:30
Raphael Dürscheid
aa2abc50bf webrtcsink: Support nvv4l2vp9enc
Naive support for nvv4l2vp9enc by assuming configuration is equivalent
to existing nvv4l2vp8enc. Validated to have relevant properties.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/983>
2022-12-02 10:18:27 +00:00
Jordan Petridis
821c23e202 net/ndi: fix build with --no-default-features
doc_show_default() is only available with gst/v1_18

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/588>
2022-11-29 21:06:12 +02:00
Vivia Nikolaidou
5bbe0eab25 ndisrc: Use actual number of channels in positions_from_mask
Otherwise it fails for mono and stereo

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/991>
2022-11-29 12:19:45 +02:00
Vivia Nikolaidou
73ce616bd9 ndisrc: Use default channel mask for audio output
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/988>
2022-11-28 17:06:07 +02:00
Sebastian Dröge
fceacf7081 Update for gst::Array / gst::List API improvements
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/985>
2022-11-27 01:12:46 +02:00
Sebastian Dröge
0e2a00cbc8 aws: Update to env_logger 0.10 for the tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/984>
2022-11-25 11:08:19 +02:00
Sebastian Dröge
456fb276d6 Revert "Update for pango API changes"
This reverts commit 6e54d3cea9.

The change was wrong and the pango bindings work the same as before
again.
2022-11-18 10:58:41 +02:00
Sebastian Dröge
6e54d3cea9 Update for pango API changes
pango::Language::from_string() can fail and also can accept None as
argument.
2022-11-18 09:46:50 +02:00
Thibault Saunier
6b11284e8a webrtcsink: Make the turn-server prop a turn-servers list
So that we can simply specify several turn servers at once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
2022-11-16 14:48:16 +00:00
Arun Raghavan
3abd13e57b aws: s3sink: Treat stopping without EOS as an error for multipart upload
This allows us to try to clean up based on configuration (abort /
complete / do nothing) if the pipeline is shut down without an EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/970>
2022-11-15 02:28:35 +00:00
Guillaume Desmottes
37cb636140 webrtc: README: fix couple of links
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/975>
2022-11-11 14:51:46 +01:00
Mathieu Duponchelle
66e7b314f7 webrtcsink: improve debug
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/972>
2022-11-10 15:00:19 +00:00
Sebastian Dröge
a5f3197651 Add missing doc features to WebRTC plugins 2022-11-07 18:06:29 +00:00
Jan Beich
9aeaac5a96 ndi: provide Unix fallback after 3fe9e4a207
error[E0425]: cannot find value `LIBRARY_NAME` in this scope
   --> net/ndi/src/ndisys.rs:336:23
    |
336 |             path.push(LIBRARY_NAME);
    |                       ^^^^^^^^^^^^ not found in this scope

error[E0425]: cannot find value `LIBRARY_NAME` in this scope
   --> net/ndi/src/ndisys.rs:339:33
    |
339 |             path::PathBuf::from(LIBRARY_NAME)
    |                                 ^^^^^^^^^^^^ not found in this scope
2022-11-05 02:51:28 +00:00
Arun Raghavan
54c84a7211 aws: Skip s3 test on Windows until we figure out why it times out 2022-11-02 13:14:08 -04:00
Sebastian Dröge
a8250abbf1 Fix various new clippy warnings 2022-11-01 10:27:48 +02:00
Sebastian Dröge
976ae5707e webrtc: Update to human_bytes 0.4 2022-10-31 14:11:29 +02:00
Sebastian Dröge
6ceeadc0f0 aws: Update to aws 0.21/0.51 2022-10-31 14:11:29 +02:00
Sebastian Dröge
ce166b4d8f whipsink: Add object to debug logs 2022-10-26 16:20:26 +03:00
Guillaume Desmottes
d46857d3b1 aws: fix title in README
The title was not matching the actual plugin name which was confusing.
2022-10-26 11:13:47 +02:00
Sebastian Dröge
bf6bdab80c webrtc: Remove version requirement from internal crate dependencies 2022-10-24 19:50:24 +03:00
Sebastian Dröge
f2223cf2cb Update versions to 0.10.0-alpha.1 2022-10-24 19:31:19 +03:00
Sebastian Dröge
b64f951160 Update to async-tungstenite 0.18 2022-10-24 18:03:33 +03:00
Sebastian Dröge
9a68f6e221 Move from imp.instance() to imp.obj()
It's doing the same thing and is shorter.
2022-10-23 23:08:46 +03:00
François Laignel
86776be58c Remove & for obj in log macros
This is no longer necessary.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1137
2022-10-23 21:22:31 +02:00
Sebastian Dröge
f045099fc1 Fix GObject type names, GStreamer debug category names and element factory names
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/198
2022-10-23 20:46:08 +03:00
Sebastian Dröge
5d44e0eb3c rtp: Move GCC bandwidth estimation element from webrtc to rtp plugin 2022-10-23 20:25:08 +03:00
Sebastian Dröge
20ad9175d8 Make GStreamer plugin/crate/library/directory names and descriptions consistent
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/238
2022-10-23 20:25:08 +03:00
Sebastian Dröge
45168639e9 Rename rtpav1 plugin to just rtp
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/243
2022-10-23 20:04:43 +03:00
Sebastian Dröge
f058a5e229 Various minor cleanups 2022-10-22 19:50:24 +03:00
François Laignel
6319d104a8 Take advantage of Into<Option<_>> args
Commit 24b7cfc8 applied changes related to nullability as declared
by gir. One consequence was that some functions signature ended up
requiring users to pass `Some(val)` when they could use `val`
before.

This commit applies changes on `gstreamer-rs` which, will honoring
the nullability stil allow users to pass `val` for the few affected
functions.

This commit also fixes the signature for `Element::request_new_pad`
which was updated upstream.
2022-10-21 11:54:24 +02:00
Sebastian Dröge
7b5d887c5b onvifmetadatacombiner: On timeout don't wait for metadata to arrive anymore but output the current video frame
Otherwise it will be too late downstream.
2022-10-21 07:08:46 +00:00
Sebastian Dröge
09ffeaf04e onvifmetadatacombiner: Add a lot of trace debug output 2022-10-21 07:08:46 +00:00
Thibault Saunier
5c89c3db69 webrtc: Rename and add to meson build the signalling server
The binary was only called `server` it has been renamed to
`gst-webrtc-signalling-server` and is installed in meson.
2022-10-20 18:20:15 +00:00
Thibault Saunier
cbdd3a7f26 webrtc: Enhance documentation 2022-10-20 12:04:43 +00:00
Sebastian Dröge
c0bf05d4bb webrtc: Minor cleanup 2022-10-20 13:20:32 +03:00
Thibault Saunier
71ed04d89b webrtc: Rename signaller and protocol crates 2022-10-20 13:32:31 +02:00
Thibault Saunier
25bda89ac8 webrtc: Update an unify rust-version and edition
So it all matches the rest of the plugins
2022-10-20 13:32:31 +02:00
Thibault Saunier
4942a916a8 webrtc: Uniformise GType names 2022-10-20 13:32:31 +02:00
Thibault Saunier
37c0239aff webrtc: Port to new ElementBuilder API 2022-10-20 13:32:31 +02:00
Thibault Saunier
ad78936365 webrtc: Enable more documentation 2022-10-20 13:32:31 +02:00
Thibault Saunier
0f0dec7fa9 webrtc: Fix fmt issues 2022-10-20 11:51:59 +02:00
Thibault Saunier
5ab7be6124 webrtc: Add SDPX license header on every file 2022-10-20 11:51:58 +02:00
Thibault Saunier
39c0dcb0d4 Plug webrtc in 2022-10-20 11:51:58 +02:00
Thibault Saunier
b164daf510 webrtc: Fix clippy issues 2022-10-20 11:51:58 +02:00
Thibault Saunier
87fd49a9bf webrtc:signalling: Remove short option for 'host' in the cli
It clashes with `--help`
2022-10-20 11:51:58 +02:00
Thibault Saunier
eb9d0bb824 Merge 'webrtcsink' from 020c7e2900 2022-10-20 11:51:58 +02:00
Sebastian Dröge
12400b6b87 Update everything for element factory builder API changes
And set properties as part of object construction wherever it makes
sense.
2022-10-19 19:43:29 +03:00
Sebastian Dröge
9ce8e93c63 rtpav1pay: Track last known upstream PTS/DTS in case not all OBUs are properly timestamped 2022-10-19 15:42:48 +03:00
Sebastian Dröge
36861edf9a rtpav1pay: Use a VecDeque instead of a Vec for the queued OBUs
And use a `Vec` plus offset for consuming partial byte buffers.
Removing the first element from a `Vec` repeatedly is not very cheap.

Also simplify calculation of the current packet by removing a mostly
unused type and keeping track of the calculations always locally instead
of sometimes storing it in the element state.
2022-10-19 15:23:10 +03:00
Sebastian Dröge
24b7cfc841 Update for GStreamer API changes 2022-10-18 19:26:52 +03:00
Arun Raghavan
03b03fe2dd whipsink: Log error body along with status code when POST fails 2022-10-18 17:01:36 +02:00
Thibault Saunier
5e7537953c webrtc: Move to net/webrtc 2022-10-18 15:18:53 +02:00
Sanchayan Maity
c63307e6d7 net/webrtc-http: whipsink: Return a proper error message & not panic
On a server error, we currently crash and panic. Return a proper error
message instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/914>
2022-10-18 10:38:57 +00:00
François Laignel
8011eadfd2 Use new format constructors
See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1128
2022-10-18 10:36:59 +00:00
Arun Raghavan
e66378d254 aws: Add a test for s3src/s3sink
This does rely on AWS credentials being provided in the environment, but
the test will be ignored if those are missing.
2022-10-18 09:51:34 +00:00
Sebastian Dröge
e17688a2da Update for pango API changes 2022-10-17 20:02:02 +03:00
Vivia Nikolaidou
0ab965335f onvifmetadataoverlay, cea608overlay: Fix pangocairo::FontMap::new()
It doesn't return an Option anymore.
2022-10-14 18:12:33 +03:00
Vivia Nikolaidou
f11b0fa5eb plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder
Simplify caps creation code
2022-10-13 19:24:57 +00:00
Sebastian Dröge
862c2af1d9 ndi: Remove unnecessary explicit Send+Sync impls
These are automatically available now.
2022-10-13 17:54:08 +00:00
Vivia Nikolaidou
dbd5a44b90 hlssink3: Use #[cfg(feature = "doc")] on gst::prelude import
It otherwise gives a warning about the unused import
2022-10-13 14:22:36 +03:00
Sebastian Dröge
5f19639d0f ndi: Various code cleanup 2022-10-13 08:52:52 +00:00
Sebastian Dröge
b2ddb34258 onvif: Switch from minidom to xmltree for parsing ONVIF timed metadata
minidom doesn't handle various valid but suboptimal XML documents.
2022-10-12 21:00:13 +00:00
Sebastian Dröge
97e0852156 ndi: Add NDI plugin to the docs 2022-10-12 22:25:13 +03:00
Sebastian Dröge
53b02a82ae ndi: Re-organize code a bit and don't make internal modules public 2022-10-12 22:09:56 +03:00
Sebastian Dröge
0a2e6e47c9 ndi: Silence some more clippy warnings 2022-10-12 22:09:55 +03:00
Sebastian Dröge
db8037d16c ndi: Update for pad default functions API changes 2022-10-12 22:09:55 +03:00
Sebastian Dröge
3fe9e4a207 ndi: Implement dynamic loading of the NDI SDK
And build the plugin on the CI and via meson.
2022-10-12 22:09:53 +03:00
Sebastian Dröge
16c036e2cc ndi: Make element factory details and debug categories more consistent 2022-10-12 21:29:07 +03:00
Sebastian Dröge
907910329f ndi: Prefix GType names with Gst 2022-10-12 21:29:07 +03:00
Sebastian Dröge
047f990c78 ndi: Integrate into the build system 2022-10-12 21:29:07 +03:00