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Author SHA1 Message Date
Guillaume Desmottes
371ac83169 fmp4: dash_vod example: use dash-mpd to generate the manifest 
Maybe a bit overkill for such simple example but more exemplary for
actual applications.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1177>
 2023-04-13 09:50:11 +02:00
Loïc Le Page
dba91bceca webrtc: fix documentation after signaller interface changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1175>
 2023-04-12 20:19:22 +02:00
Thibault Saunier
8f2273328b webrtcsrc: Return bool en 'end-session' as required 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1172>
 2023-04-12 12:17:56 +00:00
Sebastian Dröge
5dcdf645d6 net: ndi: Update to libloading 0.8 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1173>
 2023-04-12 11:03:05 +03:00
Sebastian Dröge
011f0d5fee deny: Update for miniz_oxide dependency duplication 2023-04-11 11:14:08 +03:00
Mathieu Duponchelle
355f925954 tttocea608: specify raw 608 field 
The element can only output field=0 raw 608 data.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1166>
 2023-04-11 09:26:24 +10:00
Mathieu Duponchelle
f366c20869 awstranscriber: fix what we send over for translations 
Prior to this commit, we were sending over words concatenated together
with no separators, for instance "Idon'twanttobeanemperor".

The translation service seems clever enough to translate the contents
anyway, but there is no reason to make its task harder than necessary,
and it didn't re-add separators when the target language was the same as
the source language, which resulted in less than ideal output.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
 2023-04-10 20:47:12 +00:00
Mathieu Duponchelle
408fd2030c awstranscriber: slight debug improvement 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
 2023-04-10 20:47:12 +00:00
Mathieu Duponchelle
4fcbb6ae61 textwrap: add some logs 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1171>
 2023-04-10 20:47:12 +00:00
Guillaume Desmottes
3eca8c60e3 ci: check for typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
 2023-04-10 13:35:32 +02:00
Guillaume Desmottes
403004a85e fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
 2023-04-10 13:35:32 +02:00
Mathieu Duponchelle
a455819871 webrtcsink: fix tracking of signaller state 
For the signaller to get stopped, we need to remember that we started it
in the first place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
3368f55a88 webrtcsink: don't return value from error closure 
the signal doesn't expect a return value, which meant we were panicking
as soon as the signaller tried to report an error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1167>
 2023-04-10 07:58:10 +03:00
Mathieu Duponchelle
58c8c0edc7 webrtc: signaller iface: fix session-ended vs end-session confusion 
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>
 2023-04-10 07:58:10 +03:00
Tim-Philipp Müller
0b5cf4e5fd ci: add check for symlinks 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1169>
 2023-04-08 23:20:54 +01:00
Tim-Philipp Müller
7c30430320 webrtc-api: replace LICENSE file symlink with copy 
As in !1157

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1169>
 2023-04-08 17:22:37 +01:00
Seungha Yang
6e36e2ddfd transcriberbin: Allow video with ANY caps features 
transcriberbin does not read/write video buffers actually.
Allow ANY caps features in order to avoid unnecessary GPU
upload/download

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1165>
 2023-04-08 02:40:49 +09:00
Mathieu Duponchelle
c846147275 transcriberbin: require final framerate from cea608mux 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1164>
 2023-04-07 17:46:45 +02:00
Matthew Waters
c141a82dfb webrtcsink: update docs for property and signal changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
e69b4b7f45 webrtc/signaller/iface: give variables appropriate names 
Rather than arg0, arg1, etc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
4f4e5f0d75 webrtcsink/signaller: don't call signals while having state/settings locked 
It is a recipe for deadlocks if the signal callback calls back into
webrtcsink in some way.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
1c61e46f37 webrtcsink: privatise signalling functions 
The functionality is now access through the relevant signals instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
2ac560975c webrtc/signaller: emit the relevant signals instead of the interface vtable 
In order to support the use case of an external user providing their own
signalling mechanism, we want the signals to be used and only if nothing
is connected, fallback to the default handling.  Calling the interface
vtable directly will bypass the signal emission entirely.

Also ensure that the signals are defined properly for this case. i.e.
1. Signals the the application/external code is expected to emit are
   marked as an action signal.
2. Add accumulators to avoid calling the default class handler if
   another signal handler is connected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
343b659755 webrtc/signaller: remove SignallableImplExt 
This pattern is used for subclassing and calling parent class/interface functions.
However that is not useful for the signaller object.
1. The signals are the API contract and should instead be used by
   webrtcsrc/sink to ask or provide outside for/with information.
2. The default case (no signal attached)is instead handled by default class
   handlers that call directly using the relevant rust trait.  No parent
   (GObject) vfuncs necessary.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Matthew Waters
b6e78b5f04 webrtcsink: expose signaller as a property 
in the process move the signaller field to the settings struct

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:58:13 +10:00
Thibault Saunier
8236f3e5e7 webrtcsink: Port to the 'webrtcsrc' signaller object/interface 
With contributions from:
Matthew Waters <matthew@centricular.com>

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1141>
 2023-04-07 09:03:47 +10:00
Seungha Yang
538e2e0c9e transcriberbin: Add support for runtime translation-languages update 
Allows updating translation-languages at runtime

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1162>
 2023-04-06 21:43:04 +09:00
Seungha Yang
65c6117962 transcriberbin: Wrap conversion channel elements into bin 
Make dynamic reconfiguration easier

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1162>
 2023-04-06 11:49:36 +00:00
Sebastian Dröge
884a8a8b23 meson: Update version 2023-04-06 11:25:33 +03:00
Sebastian Dröge
0bc9718e3b Update CHANGELOG.md for 0.10.6 2023-04-06 11:25:22 +03:00
Seungha Yang
762fb86ce7 awstranscriber: Reset start_time per task 
Otherwise wrong start time can be assigned if the element is
reused with state change

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1159>
 2023-04-05 18:22:59 +00:00
Sebastian Dröge
9cb211470f ndisrc: Fix copying of raw video frames with different NDI/GStreamer strides 
And also don't copy each line twice for single-plane formats.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1158>
 2023-04-05 16:45:48 +03:00
Sebastian Dröge
e549f5c4a9 deny: Update for older versions of the windows bindings 2023-04-05 12:04:21 +03:00
Matthew Waters
a8b46f1bf4 closedcaption: add cea608tocea708 element 
Implement an element that can take an input 608 caption stream and
generate a valid 708 caption stream by parsing the 608 data and
generating the equivalent DTVCCPackets and Service blocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1112>
 2023-04-05 13:00:32 +10:00
Matthew Waters
c0dc6eb35c cea608utils: track last channel for characters without channel embedded 
If a basicna character is received, it will always have a channel of 0
even if it's directed at a different data channel.  Fix by keeping track
of the last channel from other commands and using that when producing
text in the basicna subset.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1112>
 2023-04-05 12:18:15 +10:00
Matthew Waters
9a5e5db271 closedcaption: move 608 utility functions to a separate file 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1112>
 2023-04-05 12:18:15 +10:00
Loïc Le Page
f17622a1e1 webrtc: Add gstwebrtc-api subproject in net/webrtc plugin 
This subproject adds a high-level web API compatible with GStreamer
webrtcsrc and webrtcsink elements and the corresponding signaling
server. It allows a perfect bidirectional communication between HTML5
WebRTC API and native GStreamer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/946>
 2023-04-04 16:29:44 +02:00
Tim-Philipp Müller
8845f6a4c6 git: replace LICENSE file symlinks with copies 
Git will de-duplicate the contents for us anyway, and
symlinks can cause problems with some versions of git
and also on Windows.

https://github.com/mesonbuild/meson/issues/11646
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/4326

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1157>
 2023-04-04 14:26:37 +01:00
Sebastian Dröge
722dba1203 fallbacksrc: Don't check caps when linking to the fallbackswitch 
Downstream might have different caps requirements and linking might
fail. Instead of having linking fail, give upstream an opportunity to
reconfigure and otherwise have a normal negotiation error during data
flow.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/334

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1156>
 2023-04-04 10:14:06 +00:00
Guillaume Desmottes
138c318be6 uriplaylistbin: example: add queues 
Prevent pipeline starvation with some media such as
https://assets.onestream.live/studio/Videos/1080p/osl-interval-1080p-8.mp4

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1064>
 2023-04-04 08:24:55 +00:00
Seungha Yang
8576af247b transcriberbin: Set start-time-selection=first to cea608mux 
We don't want to modify running time of caption stream

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1155>
 2023-04-03 13:32:55 +00:00
Seungha Yang
4000d60305 awstranscriber: Avoid too large initial GAP event 
Initialized GstSegment.position is always zero

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1154>
 2023-04-03 13:05:15 +00:00
Vivia Nikolaidou
c6e1efa0fe livesync: Actually assume zero upstream latency when query fails 
The code said "assuming zero" but left latency at None instead of
Some(0), failing to unwrap the value later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1153>
 2023-03-31 17:40:32 +03:00
Mathieu Duponchelle
15e1844956 webrtcsink: fix calculation of fec_ratio with multiple encoders 
In this context, the bitrate variable is for all encoders, but the
max_bitrate field is per encoder. To calculate a proper FEC ratio, we
need to scale max_bitrate to the number of encoders.

+ Also clamp the fec-percentage that we set on the transceiver for extra
  safety

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1151>
 2023-03-31 12:19:07 +00:00
Sebastian Dröge
23c165dee1 deny: Update for duplicated old dependencies in dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
 2023-03-31 09:12:26 +00:00
Sebastian Dröge
315e53f064 webrtc: Update to AWS SDK 0.55/0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
 2023-03-31 09:12:26 +00:00
Sebastian Dröge
6fe806c2b5 aws: Update to AWS SDK 0.55/0.25 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1152>
 2023-03-31 09:12:26 +00:00
Mathieu Duponchelle
8cb328b6f2 transcriberbin: add support for translations 
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>
 2023-03-29 01:58:37 +02:00
David Revay
002a70a2a4 chore(webrtcsink): fix max-bitrate blurb and nick 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
 2023-03-28 16:11:05 +11:00
Vivia Nikolaidou
7a1b2d97d4 webrtcsink: Add ice-transport-policy option 
Can be used to force relay ICE candidates, ensuring TURN server is used.
Proxy to the corresponding setting in webrtcbin,

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1143>
 2023-03-27 16:12:13 +03:00