Commit graph

94 commits

Author SHA1 Message Date
Stéphane Cerveau
68c2d27e8d fmp4mux: specify the fragment duration unit
The fragment duration is expressed in nanoseconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1348>
2023-10-04 12:47:15 +02:00
Sebastian Dröge
747d9bfc6e Update plugins cache for updated raw video caps 2023-10-03 11:12:39 +03:00
Seungha Yang
ed4181617a hlssink3: Update plugin docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1306>
2023-09-25 21:34:59 +09:00
rajneeshksoni
a7fe24a294 hlssink3: Add property track-pipeline-clock-for-pdt.
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>
2023-09-20 13:54:48 +03:00
rajneeshksoni
4be24fdcaf hlssink3: Allow adding EXT-X-PROGRAM-DATE-TIME tag.
- 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>
2023-09-20 13:54:48 +03:00
Sebastian Dröge
95a7a3c0ec gtk4: Only support RGBA/RGB in the GL code path
For all other component orderings a shader is necessary to re-order the
components for what GTK expects.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1312>
2023-09-20 13:22:41 +03:00
Seungha Yang
1de7754616 webrtcsink: Add support for d3d11 memory and qsvh264enc
Adding d3d11 memory and qsvh264enc support

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1322>
2023-09-15 00:26:04 +09:00
Sebastian Dröge
146a96686b Update docs for new order of raw video formats
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1308>
2023-08-29 19:45:04 +03:00
Taruntej Kanakamalla
de6d2e7f40 net/webrtc: rename whipwebrtcsink as whipclientsink
add a deprecation warning in whipsink to indicate it
should be used only for RTP content
add documentation in whipsink code regarding usage and
deprecation

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1282>
2023-08-26 10:53:30 +05:30
Mathieu Duponchelle
e905299eba generic: expose inter plugin
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>
2023-08-14 08:13:12 +00:00
Andoni Morales Alastruey
3000b08ec7 webrtcsrc: add support for navigation events
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>
2023-08-10 17:43:51 +00:00
Loïc Le Page
7af2ff0843 net/webrtc/signaller: advertise running producers in Listener mode
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>
2023-08-10 17:30:21 +02:00
Sebastian Dröge
983f990fe9 Update docs after GStreamer update on the CI 2023-07-06 13:48:59 +03:00
Olivier Crête
793ee66afa webrtcsink: Add LiveKit WebRTC sink and signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
2023-07-05 21:43:17 +00:00
Mathieu Duponchelle
84a33ca7b9 webrtcsink: bring in signalling code from whipsink as a signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1168>
2023-06-16 00:32:56 +02:00
Vivia Nikolaidou
063871a1eb togglerecord: Add support for non-live inputs
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>
2023-06-14 15:58:04 +03:00
Guillaume Desmottes
4683291c1f fallbackswitch: add 'stop-on-eos' property
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>
2023-06-13 14:49:06 +02:00
Sebastian Dröge
c65b3429ad Use MPL as license specifier for plugins only requiring GStreamer < 1.20
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1235>
2023-06-07 19:13:55 +03:00
Mathieu Duponchelle
fda5aed89f webrtcsink: encoded streams: address last review comments
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
2023-06-06 16:05:28 +02:00
Mathieu Duponchelle
a20855dfd9 webrtcsink: expose consumer-pipeline-created signal
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>
2023-05-25 13:15:52 +02:00
Guillaume Desmottes
7ebf2d7a4f fallbackswitch: document the pad priority ordering
I just wasted lots of time trying to figure out why my higher priority
pad wasn't used...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1208>
2023-05-15 16:13:20 +02:00
Seungha Yang
773fcd0780 transcriberbin: Add "language-code" property
Proxy the child transcriber element's property so that transcriberbin
can apply the property with required state management

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1205>
2023-05-10 19:12:01 +00:00
François Laignel
680d5221db net/webrtc: src: add signal "request-encoded-filter"
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
2023-05-09 12:02:15 +02:00
Arun Raghavan
aabfb61834 ffv1dec: Drop rank for now
We'll keep the rank lower than avdec_ffv1, at least until we're
comfortable with support for the entire range of possible inputs working
well.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1174>
2023-04-13 15:58:49 +00:00
Mathieu Duponchelle
f1fd8d84c3 webrtc: extract a BaseWebRTCSink
For documentation purposes, AwsKVSWebRTCSink should not inherit from
another element.

+ Mark base class as plugin API and update plugin cache

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1178>
2023-04-13 15:06:59 +00:00
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
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
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
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
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
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
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
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
François Laignel
162db2f3b9 net/aws/transcriber: fix translate lookahead
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16 12:39:15 +01:00
François Laignel
d5d6a4daf9 net/aws/transcriber: rename prop transcript-lookahead & TranslationSrcPad
... as translate-lookahead and TranslateSrcPad.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
2023-03-16 12:37:31 +01:00
François Laignel
299e25ab3c net/aws/transcriber: translate: optional experimental translation tokenization
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>
2023-03-14 13:48:32 +00:00
François Laignel
743e97738f net/aws/transcriber: add translation request src pads
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>
2023-03-14 13:48:32 +00:00
Mathieu Duponchelle
584392049c net/webrtc: implement AWS KVS signaller
And expose a wrapper webrtcsink variant, aws-kvs-webrtcsink.

This adds support in webrtcsink for processing a consumer offer, instead
of producing one.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1114>
2023-03-09 15:39:09 +00:00
François Laignel
00153754bb net/aws: use aws-sdk-transcribestreaming
Switch from manual webservice client impl to `aws-sdk-transcribestreaming`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
2023-03-01 08:47:58 +00:00
Thibault Saunier
ce3bb2f1d4 Add a webrtcsrc element
Updating the docker image to include:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3236

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
2023-02-28 20:50:15 -03:00
Arun Raghavan
39e0acb55a hlssink3: Fix case on unspecified playlist type nick for consistency
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1089>
2023-02-10 23:07:12 +00:00
rajneeshksoni
994c79569e awss3sink: Add properties to set content-Type and content-disposition.
for uploaded object default content-type is set to binary/octet-stream,
which is correct.
metadata cannot be used to set content-type and content-disposition as
setting metadata add a prefix x-amz-meta to key
e.g. setting metadate "content-type=video/mp4" actually set value as
x-amz-meta-content-type. So these has to be seaprate property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1085>
2023-02-09 19:04:07 +00:00
Simon Himmelbauer
3c31c98d95 spotifyaudiosrc: Support configurable bitrate
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1073>
2023-02-09 00:02:30 +02:00
rajneeshksoni
0f383a6545 hlssink3: Allow setting i-frame-only playlist.
HLS allows manifest where all segments are single ifames.
This manifest requires `EXT-X-I-FRAMES-ONLY` tag in the
manifest.
I-FRAMES-ONLY playlist segments are video only segments.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1070>
2023-02-08 14:04:46 +00:00
Sebastian Dröge
6f26e3bf79 mp4/fmp4: Update docs
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1075>
2023-02-04 16:32:17 +02:00
Sebastian Dröge
5506f8001e rtpav1pay: Add support for tu/frame aligned input
In this case every buffer can be sent out immediately and makes up a
whole frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
d6cb9d72d8 rtpav1depay: Don't output full TUs but just OBUs as they come
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>
2023-02-02 20:24:27 +02:00
Sebastian Dröge
2a3d962dc5 fmp4mux: Add support for sub-fragments / chunking
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>
2023-01-27 19:28:27 +00:00