webrtcsink was starting the negotiation process on Ready and concurrently
moving the consumer pipeline to Playing, but when answering the remote
description was set so fast that input streams were connected (and the time
format set on appsrc) before the state change to Paused had completed.
This meant gst_base_src_start was happening after that and setting the format
back to bytes, the time segment that was next coming in then caused:
basesrc gstbasesrc.c:4255:gst_base_src_push_segment:<video_0> segment format mismatched, ignore
And the consumer pipeline errored out.
The same issue existed in theory when webrtcsink was creating the offer,
but was much harder to trigger as it required that the remote answer
came in before the state change to Paused had completed.
This commit fixes the issue by simply waiting for the state to have
changed to Paused before negotiating.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1730>
This is one example of how a consumer might send over custom upstream
event requests to the producer.
As webrtcsink will deserialize numbers in priority as integers, we need
a custom stringifying function to ensure members of the matrix array are
indeed serialized with the floating point.
An optional stringifier parameter is thus added to the
sendControlRequest API.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
.. and deprecate data channel navigation in favor of it.
A new property, "enable-data-channel-control" is exposed, when set to
TRUE a control data channel is offered, over which can be sent typed
upstream events.
This means further upstream events will be usable, for now only
navigation and custom upstream events are handled.
In addition, send response messages to notify the consumer of whether
its requests have been handled.
In the future this can also be extended to allow the consumer to send
queries, or seek events ..
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
This commit adds support for raw payloads such as L24 audio to `webrtcsink` &
`webrtcsrc`.
Most changes take place within the `Codec` helper structure:
* A `Codec` can now advertise a depayloader. This also ensures that a format
not only can be decoded when necessary, but it can also be depayloaded in the
first place.
* It is possible to declare raw `Codec`s, meaning that their caps are compatible
with a payloader and a depayloader without the need for an encoder and decoder.
* Previous accessor `has_decoder` was renamed as `can_be_received` to account
for codecs which can be handled by an available depayloader with or without
the need for a decoder.
* New codecs were added for the following formats:
* L24, L16, L8 audio.
* RAW video.
The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw
audio or video.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
We can use `is_some_and(...)` instead of `map_or(false, ...)`.
Also in a few places the factory was retrieved multiple times, one time
with unwrapping and another time with handling the `None` case
correctly. Instead of unwrapping, move code to handle the `None` case.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
- generate a new session id for every new client
use the session id in the resource url
- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer
- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
- Add a new structure Session
- manage each producer using a session
- avoid send EOS when a session terminates, instead keep running
waiting for any new producer to connect
- Maintain a bin element per session
- each session bin encapsulates webrtcbin and the decoder if needed
as well as the parser and filter if requested by the application
(through request-encoded-filter)
- this will be helpful to cleanup the session's respective elements
when the corresponding producer terminates the session
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
This commit adds an Android `webrtcsrc` based example with the following
features:
* A first view allows retrieving the producer list from the signaller (peer ids
are uuids which are too long to tap, especially using an onscreen keyboard).
* Selecting a producer opens a second view. The first available video stream is
rendered on a native Surface. All the audio streams are rendered using
`autoaudiosink`.
Available Settings:
* Signaller URI.
* A toggle to prefer hardware decoding for OPUS, otherwise the app defaults to
raising `opusdec`'s rank. Hardware decoding was moved aside since it was found
to crash the app on all tested devices (2 smartphones, 1 tv).
**Warning**: in order to ease testing, this demonstration application enables
unencrypted network communication. See `AndroidManifest.xml`.
The application uses the technologies currenlty proposed by Android Studio when
creating a new project:
* Kotlin as the default language, which is fully interoperable with Java and
uses the same SDK.
* gradle 8.6.
* kotlin dialect for gradle. The structure is mostly the same as the previously
preferred dialect, for which examples can be found online readily.
* However, JNI code generation still uses Makefiles (instead of CMake) due to
the need to call [`gstreamer-1.0.mk`] for `gstreamer_android` generation.
Note: on-going work on that front:
- https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/1466
- https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6794
Current limitations:
* x86 support is currently discarded as `gstreamer_android` libs generation
fails (observed with `gstreamer-1.0-android-universal-1.24.3`).
* A selector could be added to let the user chose the video streams and
possibly decide whether to render all audio streams or just select one.
Nice to have:
* Support for the synchronization features of the `webrtc-precise-sync-recv`
example (NTP clock, RFC 7273).
* It could be nice to use Rust for the specific native code.
[`gstreamer-1.0.mk`]: https://gitlab.freedesktop.org/gstreamer/cerbero/-/blob/main/data/ndk-build/gstreamer-1.0.mk
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1578>
If a user constrained the supported CAPS, for instance using `video-caps`:
```shell
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420 ! x264enc \
! webrtcsink video-caps=video/x-vp8
```
... a panic would occur which was internally caught without the user being
informed except for the following message which was written to stderr:
> thread 'tokio-runtime-worker' panicked at net/webrtc/src/webrtcsink/imp.rs:3533:22:
> expected audio or video raw caps: video/x-h264, [...] <br>
> note: run with `RUST_BACKTRACE=1` environment variable to display a backtrace
The pipeline kept running.
This commit converts the panic into an `Error` which bubbles up as an element
`StreamError::CodecNotFound` which can be handled by the application.
With the above `gst-launch`, this terminates the pipeline with:
> [...] ERROR webrtcsink net/webrtc/src/webrtcsink/imp.rs:3771:gstrswebrtc::
> webrtcsink:👿:BaseWebRTCSink::start_stream_discovery_if_needed::{{closure}}:<webrtcsink0>
> Error running discovery: Unsupported caps: video/x-h264, [...] <br>
> ERROR: from element /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
> There is no codec present that can handle the stream's type. <br>
> Additional debug info: <br>
> net/webrtc/src/webrtcsink/imp.rs(3772): gstrswebrtc::webrtcsink:👿:BaseWebRTCSink::
> start_stream_discovery_if_needed::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
> Failed to look up output caps: Unsupported caps: video/x-h264, [...] <br>
> Execution ended after 0:00:00.055716661 <br>
> Setting pipeline to NULL ...
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1540>
Clippy caught the missing feature `signal` which is used by the WebRTC precise
synchronization examples. When running `cargo` `check`, `build` or `clippy`
without `no-default-dependencies`, this feature was already present due to
dependents crates.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1541>
When swapping between several development branches, compilation times can be
frustrating. This commit proposes adding features to control which signaller
to include when building the webrtc plugin. By default, all signallers are
included, just like before.
Compiling the `webrtc-precise-sync` examples with `--no-default-features`
reduces compilation to 267 crates instead of 429 when all signallers are
compiled in.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1539>
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization)
for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to
identify the clock. These attributes are handled by `rtpjitterbuffer` on the
consumer side. They MUST be part of the SDP offer.
When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the
clock's origin. Because the payloaders are not instantiated when the offer is
sent to the consumer, the RTP offset is set to 0 and the payloader
`timstamp-offset`s are set accordingly when they are created.
The `webrtc-precise-sync` examples were updated to be able to start with an NTP
(default), a PTP or the system clock (on the receiver only). The rtp jitter
buffer will synchronize with the clock signalled in the SDP offer provided the
sender is started with `--do-clock-signalling` & the receiver with
`--expect-clock-signalling`.
[RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
When an encoder was not supported by the `VideoEncoder` `bitrate` accessors, an
`unimplemented` panic would occur which would poison `state` & `settings`
`Mutex`s resulting in other threads panicking, notably entering `end_session()`,
which lead to many failures in `BinImplExt::parent_remove_element()` until a
segmentation fault ended the process. This was observed using `vaapivp9enc`.
This commit logs a warning if an encoder isn't supported by the `bitrate`
accessors and silently by-passes `bitrate`-related operations when unsupported.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1534>
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/1520>
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/1511>
Instead of exposing all ids properties as strings, we now have two
signaller implementations exposing those properties using their actual
type. This API is more natural and save the element and application
conversions when using numerical ids (Janus's default).
I also removed the 'joined-id' property as it's actually the same id as
'feed-id'. I think it would be better to have a 'janus-state' property or
something like that for applications willing to know when the room has
been joined.
This id is also no longer generated by the element by default, as Janus
will take care of generating one if not provided.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1486>
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/1482>
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>
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>
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>
