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Author SHA1 Message Date
Martin Nordholts
0faa74b74d rtpgccbwe: Add increasing_duration and counter to existing gst::log!() 
Add `self.increasing_duration` and `self.increasing_counter`
to logs to provide more details of why `overuse_filter()`
determines overuse of network.

To get access to the latest values of those fields we need
to move down the log call. But that is fine, since no other
logged data is modified between the old and new location of
`gst::log!()`.

We do not bother logging `self.last_overuse_estimate` since
that is simply the previously logged value of `estimate`. We
must put the log call before we write the latest value to it
though, in case we want to log it in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:18:03 +03:00
François Laignel
d0461b190f aws: improve error message logs 
The `Display` and `Debug` trait for the AWS error messages are not very useful.

- `Display` only prints the high level error, e.g.: "service error".
- `Debug` prints all the fields in the error stack, resulting in hard to read
  messages with redudant or unnecessary information. E.g.:

> ServiceError(ServiceError { source: BadRequestException(BadRequestException {
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), meta: ErrorMetadata { code: Some("BadRequestException"),
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), extras: Some({"aws_request_id": "1b8bbafd-5b71-4ba5-8676-28432381e6a9"}) } }),
> raw: Response { status: StatusCode(400), headers: Headers { headers:
> {"x-amzn-requestid": HeaderValue { _private: H0("1b8bbafd-5b71-4ba5-8676-28432381e6a9") },
> "x-amzn-errortype": HeaderValue { _private:
> H0("BadRequestException:http://internal.amazon.com/coral/com.amazonaws.transcribe.streaming/") },
> "date": HeaderValue { _private: H0("Tue, 26 Mar 2024 17:41:31 GMT") },
> "content-type": HeaderValue { _private: H0("application/x-amz-json-1.1") },
> "content-length": HeaderValue { _private: H0("315") }} }, body: SdkBody {
> inner: Once(Some(b"{\"Message\":\"1 validation error detected: Value 'test'
> at 'languageCode' failed to satisfy constraint: Member must satisfy enum value
> set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT,
> es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP,
> ca-ES, es-US, fr-CA, en-GB]\"}")), retryable: true }, extensions: Extensions {
> extensions_02x: Extensions, extensions_1x: Extensions } } })

This commit adopts the most informative and concise solution I could come up
with to log AWS errors. With the above error case, this results in:

> service error: Error { code: "BadRequestException", message: "1 validation
> error detected: Value 'test' at 'languageCode' failed to satisfy constraint:
> Member must satisfy enum value set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI,
> pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN,
> en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US, fr-CA, en-GB]",
> aws_request_id: "a40a32a8-7b0b-4228-a348-f8502087a9f0" }

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:17:56 +03:00
François Laignel
85ce07d184 aws: use fixed BehaviorVersion 
Quoting [`BehaviorVersion` documentation]:

> Over time, new best-practice behaviors are introduced. However, these
> behaviors might not be backwards compatible. For example, a change which
> introduces new default timeouts or a new retry-mode for all operations might
> be the ideal behavior but could break existing applications.

This commit uses `BehaviorVersion::v2023_11_09()`, which is the latest
major version at the moment. When a new major version is released, the method
will be deprecated, which will warn us of the new version and let us decide
when to upgrade, after any changes if required. This is safer that using
`latest()` which would silently use a different major version, possibly
breaking existing code.

[`BehaviorVersion` documentation]: https://docs.rs/aws-config/1.1.8/aws_config/struct.BehaviorVersion.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:17:49 +03:00
Sebastian Dröge
b05436d6d3 reqwest: Update to reqwest 0.12 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:17:31 +03:00
Philippe Normand
23403b5c9a Fix clippy warnings after upgrade to Rust 1.77 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1531>
 2024-04-08 15:15:26 +03:00
François Laignel
fd154d272c webrtcsrc: add do-retransmission property 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
François Laignel
ede354d7a5 webrtcsink: prevent video-info error log for audio streams 
The following error is logged when `webrtcsink` is feeded with an audio stream:

> ERROR video-info video-info.c:540:gst_video_info_from_caps:
>       wrong name 'audio/x-raw', expected video/ or image/

This commit bypasses `VideoInfo::from_caps` for audio streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
François Laignel
158fe80779 rtp: gccbwe: don't break downstream assumptions pushing buffer lists 
Some elements in the RTP stack assume all buffers in a `gst::BufferList`
correspond to the same timestamp. See in [`rtpsession`] for instance.
This also had the effect that `rtpsession` did not create correct RTCP as it
only saw some of the SSRCs in the stream.

`rtpgccbwe` formed a packet group by gathering buffers in a `gst::BufferList`,
regardless of whether they corresponded to the same timestamp, which broke
synchronization under certain circonstances.

This commit makes `rtpgccbwe` push the buffers as they were received: one by one.

[`rtpsession`]: bc858976db/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c (L2462)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
Guillaume Desmottes
a502dba6d5 webrtc: janus: handle 'hangup' messages from Janus 
Fix error about this message not being handled:

{
   "janus": "hangup",
   "session_id": 4758817463851315,
   "sender": 4126342934227009,
   "reason": "Close PC"
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
Guillaume Desmottes
be055f6dfa webrtc: janus: handle 'destroyed' messages from Janus 
Fix this error when the room is destroyed:

ERROR   webrtc-janusvr-signaller imp.rs:413:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x55b166a3fe40> Unknown message from server: {
   "janus": "event",
   "session_id": 6667171862739941,
   "sender": 1964690595468240,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "destroyed",
         "room": 8320333573294267
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
Sebastian Dröge
c982db73a7 rtp: Switch from chrono to time 
Which allows to simplify quite a bit of code and avoids us having to
handle some API deprecations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-21 13:30:20 +02:00
Guillaume Desmottes
21f59c65da webrtc: allow resolution and framerate input changes 
Some changes do not require a WebRTC renegotiation so we can allow
those.

Fix #515

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
7a90e96332 livekit_signaller: Added missing getter for excluded-producer-peer-ids 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
283d1568b4 webrtcsrc: Removed incorrect URIHandler from LiveKit source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
51287705ce livekit_signaller: Improved shutdown behavior 
Without sending a Leave request to the server before disconnecting, the
disconnected client will appear present and stuck in the room for a little
while until the server removes it due to inactivity.

After this change, the disconnecting client will immediately leave the room.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
e9edee131b webrtcsrc: Removed flag setup from WhipServerSrc 
It's already done in the base class

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
feb01510f9 webrtcsrc: Updated readme for LiveKit source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
32e13f0a10 webrtcsrc: Added LiveKit source element 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
c8dcd50846 webrtcsink: Updated livekitwebrtcsink for new signaller constructor 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Jordan Yelloz
59ee2721bf livekit_signaller: Added dual-role support 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:24:55 +01:00
Guillaume Desmottes
133b527391 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/1508>
 2024-03-20 16:54:45 +02:00
Guillaume Desmottes
7f460c2db8 webrtc: janus: room id not optional in 'joined' message 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:54:17 +02:00
Guillaume Desmottes
8361471fcc 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/1508>
 2024-03-20 16:54:02 +02:00
Guillaume Desmottes
b9ea05a14a webrtc: janus: numerical room ids are u64 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1508>
 2024-03-20 16:53:51 +02:00
Jordan Yelloz
048d51d9d9 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/1508>
 2024-03-20 16:53:22 +02:00
Xavier Claessens
7edf94f98b 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/1476>
 2024-02-26 14:24:29 +02:00
Xavier Claessens
ea59544c71 janusvr: Room IDs can be strings 
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
 2024-02-26 14:24:25 +02:00
Maksym Khomenko
4e86b0f3c8 webrtcsink: extensions: separate API and signal checks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
 2024-02-26 14:24:02 +02:00
Maksym Khomenko
98411e97f1 webrtcsink: apply rustfmt 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1476>
 2024-02-26 14:23:56 +02:00
Xavier Claessens
3a0f30be96 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/1476>
 2024-02-26 14:23:45 +02:00
Sebastian Dröge
3679db5740 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/1476>
 2024-02-26 14:20:53 +02:00
Jordan Yelloz
606352d7cf 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/1460>
 2024-02-12 18:11:42 +02:00
Sebastian Dröge
aa2d056ea1 Update to async-tungstenite 0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1460>
 2024-02-12 18:11:31 +02:00
Sebastian Dröge
16e001e3f2 Update dependency versions for gtk-rs-core / gtk4-rs / gstreamer-rs and local crates 2024-02-08 19:40:08 +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