Previously livesync was waiting for the start timestamp of the current
buffer after looking at the queue and right before pushing it
downstream. This meant that it generally looked too early at the queue
and especially that upstream had to provide the next buffer already at
the start timestamp of the previous one.
Instead, now wait before looking at the queue and wait for the end
timestamp of the previous buffer. Once the previous buffer has expired,
a new buffer really needs to be available or otherwise a filler buffer
has to be pushed downstream.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1250>
I've looked at the GstQueue code again and tried making livesync behave
better with EOS. This isn't very well tested, though. My goal was to
make this look saner but I think this should be reviewed by someone who
knows the queue code.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1251>
Live input + is-live=false:
While not recording, drop input
When recording is started, offset to collapse the gap
Live input + is-live=true:
While not recording, drop input
Don't modify the offset
Non-live input + is-live=false:
While not recording, block input
Don't modify the offset
Non-live input + is-live=true:
While not recording, block input
When recording is started, offset to current running time
Co-authored-by: Jan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1206>
Fix the following use case:
- main input of fallbackswitch is finite (a media file)
- fallback input is infinite (videotestsrc)
- main input is done and send eos, which is propagated downstream
- fallbackswitch switches to fallback, sending STREAM_START which reset
EOS downstream (aggregator does that)
- fallback input keeps pushing buffers forever.
Solve it by adding a 'stop-on-eos' property so fallbackswitch stops
pushing property once the main input is eos.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1242>
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>
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>
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>
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>
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>
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>
