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>
Implement new signaller WhipServerSignaller
- an http server using 'warp'
- handlers for the POST, OPTIONS, PATCH and DELETE
- fixed path `/whip/endpoint` as the URI
- fixed value 'whip-client' as the producer peer id
- fixed resource url `/whip/resource/whip-client`
Derive whipserversrc element from BaseWebRTCSrc
- implement constructed method for ObjectImpl to set
non-default signaller, i.e., WhipServerSignaller
- bind the properties stun-server and turn-servers to those on
the Signaller
Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller
- it will be emitted by the webrtcsrc when the webrtcbin element is ready
- the closure for this signal will in turn connect to webrtcbin's ice-gathering-state
and perform send with the answer sdp via the channel
- the WhipServer will hold its HTTP response in POST handler until this signal
is received or timeout which happens early
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
add a new signal webrtcbin-ready in this place doing same
thing but can be used for both consumers and producers
Please note this change is only to the consumer-added
signal on the signaller interface.
The consumer-added signal on the webrtcsink is unchanged
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
The "signaller" property used to be defined as MUTABLE_READY which meant that
the property was always set after `constructed()` was called.
Since `connect_signaller()` was called from `constructed()`, only the default
signaller was used.
This commit sets the "signaller" property as CONSTRUCT_ONLY. Using a builder,
this property will now be set before the call to `constructed()`.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1324>
This is required to take care of clock skew between
system time and pipeline time.
`track-pipeline-clock-for-pdt: true` mean utd time is
sampled for first segment and for subsequent segments
keep adding the time based on pipeline clock. difference
of segment duration and PDT time will match.
track-pipeline-clock-for-pdt: false` mean utd time is
sampled for each segment. system time may jump forward
or backward based on adjustments. If application needs
to synchronization of external events `false` is
recommended.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1145>
- connect to `format-location-full` it provide the first
sample of the fragment. preserve the running-time of the
first sample in fragment.
- on fragment-close message, find the mapping of running-time
to UTC time.
- on each subsequent fragment, calculate the offset of the
running-time with first fragment and add offset to base
utc time
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1145>
This new plugin exposes two elements, intersink and intersrc. These act
as wormholes for data in the same process and can be used to forward
data from one pipeline to another.
The implementation makes use of gstreamer-utils' StreamProducer, and
supports dynamically adding and removing consumers, before and after
producers, and changing producer names while PLAYING, both on the sink
and the src.
This initial implementation comes with a small demo, and a few tests.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1257>
This provides support GstNavigation events handling in webrtcsrc so that
a GStreamer client can be used to control remotely a GStreamer server,
similar to how the web client is capable of controlling a wpesrc.
This is part of a larger set of patches that require more work on the
sinks and sources.
server: d3d11screencapturesrc ! webrtcsink enable-data-channel-navigation=true
client: webrtcsrc enable-data-channel-navigation=true ! d3d11videosink
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1281>
When starting a webrtcsrc-signaller client in Listener mode, only the producers
started after the client connection were advertised. All currently
running producers were ignored unlike the gstwebrtc-api behavior. This
commit now lists all running producers when the client Listener connects
and advertises them through the "producer-added" signal.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1296>
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>
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>
Session ending is bidirectional: the signaller can tell the sink that a
session was ended, and the sink can tell the signaller to end a session.
As such, two signals are needed, before this patch the second case was
not working as in essence the sink was telling itself that a session was
ended, and obviously failing to even find it when trying to end it again.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
With this, if the transcriber element in use supports "translation_src_"
request source pads, the user can now specify what languages to
translate to and how to map them to 608 channels (only CC1 and CC3 are
supported).
For instance, translation-languages="languages, CC3=transcript, CC1=fr"
will cause the original transcript to be muxed into the CC3 channel, and
the French translation to be muxed into the CC1 channel.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1149>
This commit adds an optional experimental translation tokenization feature.
It can be activated using the `translation_src_%u` pads property
`tokenization-method`. For the moment, the feature is deactivated by default.
The Translate ws accepts '<span></span>' tags in the input and adds matching
tags in the output. When an 'id' is also provided as an attribute of the
'span', the matching output tag also uses this 'id'.
In the context of close captions, the 'id's are of little use. However, we can
take advantage of the spans in the output to identify translation chunks, which
more or less reflect the rythm of the input transcript.
This commit adds simples spans (no 'id') to the input Transcript Items and
parses the resulting spans in the translated output, assigning the timestamps
and durations sequentially from the input Transcript Items. Edge cases such as
absence of spans, nested spans were observed and are handled here. Similarly,
mismatches between the number of input and output items are taken care of by
some sort of reconcialiation.
Note that this is still experimental and requires further testings.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
This commit adds an optional transcript translation feature implemented as
request src Pads.
When requesting a src Pad, the user can specify the translation language code
using Pad properties 'language-code'.
The following properties are defined on the Element:
- 'transcribe-latency': formerly 'latency', defines the expected latency for
the Transcribe webservice.
- 'translate-latency': defines the expected latency for the Translate
webservice.
- 'transcript-lookahead': maximum transcript duration to send to translation
when a transcript is hitting its deadline and no punctuation was found.
When the input and output languages are the same, only the 'transcribe-latency'
is used for the Pad. Otherwise, the resulting latency is the addition of
'transcribe-latency' and 'translate-latency'.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1109>
