It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.
To solve this remember if unlock() was called and reset this in
unlock_stop().
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
- 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>
There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession. This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other. The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.
Demultiplexing happens internally instead of relying on separate
elements.
Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
Add to awss3sink and awss3putobjectsink elements the following
paramerters which are set on the uploaded S3 objects:
* cache-control;
* content-encoding; and
* content-language
Bugfix: Set the content-type and content-disposition values in the S3
putobject call. Previously the params were defined on the element but
unused.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
Unit tests specify a 0-based offset, so printing that plus the
random initial offset on failure is just needlessly confusing,
so subtract the initial offset when printing expected/actual
values. The real values are still printed as part of the assert.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
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>
Only configure header extensions from the source pad caps if they exist
already in the source pad caps, otherwise the configuration will fail.
Extensions that are added via the signals might not exist in the source
pad caps yet and would be added later.
Also, if configuring an existing extension from the new caps fails,
remove it and try to request a new extension for it.
Additionally don't remove extensions from the caps that can't be
provided. No header extensions for them would be added to the packets,
but that's not a problem. Removing them on the other hand would cause
negotiation to fail. This only affects extensions that are already
included in the caps.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
If configuring an existing extension from the new caps fails, remove it
and try to request a new extension for it.
Also remove all extensions from the list that are not provided in the
caps, instead of passing RTP packets to all of them anyway.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
In our tests, the slope (found with linear regression) on a
history of the (smoothed) accumulated inter-group delays
gives a more stable congestion control. In particular,
low-end devices becomes less sensitive to spikes in
inter-group delay measurements.
This flavour of delay based bandwidth estimation with Google
Congestion Control is also what Chromium is using.
To make it easy to experiment with the new estimator, as
well as add support for new ones in the future, also add
infrastructure for making delay estimator flavour selectable
at runtime.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
Rust targets without support for `AtomicU64` is still
somewhat common. Running
git grep -i 'max_atomic_width: Some(32)' | wc -l
in the Rust compiler repo currently counts to 34 targets.
Change the `RtpBasePay2::ssrc_collision` from `AtomicU64` to
`Mutex<Option<u32>>`. This way we keep support for these
targets.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1562>
While monitoring and debugging rtpgccbwe, it is very helpful
to get continuous values of what it considers the effective
bitrate. Right now such prints will stop coming once the
algorithm stabilizes. Print it in more places so it keeps
coming. Use the same format to make it simpler to extract
the values by parsing the logs.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1567>
When debugging rtpgccbwe it is helpful to know if it is
delay based or loss based band-width estimation that puts a
bound on the current target bitrate, so add logs for that.
To minimize the time we need to hold the state lock, perform
the logging after we have released the state lock.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1561>
This set of changes implements the below fixes:
- Allow certificates to be specified for client/quicsink
- Secure connection being true on server/quicsrc and false on
client/quicsink still resulted in a successful connection
instead of server rejecting the connection
- Using secure connection with ALPN was not working
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
