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>
Allow outputting sub-fragments (chunks in CMAF terms) that are shorter
than the fragment duration and don't usually start on a keyframe. By
this the buffering requirements of the element is reduced to one chunk
duration, as is the latency.
This is used for formats like low-latency / LL-HLS and DASH.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1059>
Required a slight rework of the build file, and how options are passed
to cargo_wrapper.py
Necessitated a bump of the required gstreamer version to 1.20, which
should be fine for the meson build since its primary function is to be
built as part of the gstreamer monorepo build to get a dev env.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1028>
It attempts to produce a (nearly) gapless live stream by synchronizing
its output to the running time and forwarding the next input buffer if
its start is (nearly) flush with the end of the last output buffer.
If the input buffer is missing or too far in the future, it duplicates
the last output buffer with adjusted timestamps. If it is operating on a
raw audio stream, it will fill duplicate buffers with silence.
If an input buffer arrives too late, it is thrown away. If the last
input buffer was accepted too long ago (according to `late-threshold`),
a late input buffer is accepted anyway, but immediately considered a
duplicate. Due to the silence-filling, this has no effect on audio, but
video gets a "slideshow" effect instead of freezing completely.
The "many-repeats" property will be notified when this element has
recently duplicated a lot of buffers or recovered from such a state.
Co-authored-by: Vivia Nikolaidou <vivia@ahiru.eu>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/708>
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>
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>
This makes it easy to generate "listenable" signals and to evaluate
discontinuities.
When the `tuning` feature is activated and the `main-elem` property
is set, the element can log the parked duration in %, which is an
image of the CPU usage for the ts-context.
This commit adds a test mode to `udpsrc-benchmark-sender` which
generates default audio buffers from `ts-audiotestsrc`. The `rtp`
mode is modified so that it uses `ts-audiotestsrc`.
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/1410
Created a new plugin 'webrtchttp' to implement all the
WebRTC HTTP protocols under /net/webrtc-http directory.
WhipSink wraps around 'webrtcbin' with HTTP capabilites
to exchange SDP offer/answer so an ICE/DTLS session can
be established between the encoder/media producer (WHIP client)
and the broadcasting ingestion endpoint (Media Server).
Once the ICE/DTLS session is set up, the media will
flow unidirectionally from the WHIP client to the
broadcasting ingestion endpoint (Media Server).
Spec:
https://www.ietf.org/archive/id/draft-ietf-wish-whip-04.html
