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Author SHA1 Message Date
Sebastian Dröge 957a28f239 webrtcsink: Set VP8/VP9 payloader based on payloader element factory name 
Instead of checking the encoder's name. There are more VP8/VP9 encoders
than the ones from the vpx plugin.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:45:17 +03:00
Mathieu Duponchelle 1dd13c4812 webrtcsink: fix session_id / peer_id confusion 
In a few places, for instance parameter names, peer_id was still used
when session_id was actually getting passed.

Go through all instances of peer_id in webrtcsink/imp.rs and address
those mix-ups.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1269>
 2023-07-07 05:33:30 +00:00
Bilal Elmoussaoui dd2d7d9215 Use re-exported once_cell 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Bilal Elmoussaoui 2cc98bf410 Adapt to glib::Continue rename 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Sebastian Dröge 58adebb325 Fix a couple of typos 2023-07-06 13:50:17 +03:00
Olivier Crête 08b6251a7a webrtc-utils: Ensure there is only one cancellable call at a time 
Since we only have one canceller at a time, panic if one try to
use it twice in parallel.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
Olivier Crête 817b60a758 webrtc: Value.get() is already type checks in the property calls 
GObject will have ensured we get a GValue of the right type.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1262>
 2023-07-05 21:43:17 +00:00
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
Seungha Yang 1f0ce101eb awstranscriber: Tone down log message 
It's not an ERROR case at all

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1263>
 2023-06-28 23:57:54 +09:00
Sebastian Dröge c350f3c2af webrtcink: Use correct property types for nvvideoconvert 
These are enums and not plain integers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1256>
 2023-06-26 14:48:58 +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
Mathieu Duponchelle f00a169081 webrtcsrc: add twcc extension to codec-preferences when present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1245>
 2023-06-15 20:41:53 +00:00
Mathieu Duponchelle 1200ae0ee6 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Mathieu Duponchelle 64056c5527 net/webrtc: improve documentation layout 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1239>
 2023-06-14 22:27:15 +02:00
Sebastian Dröge 8a7a1f519c webrtc: Update to fastrand 2 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1240>
 2023-06-09 09:36:51 +03:00
Mathieu Duponchelle 81ae675f2d webrtcsink: don't try to use cudaconvert if not present 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00
Mathieu Duponchelle 7f78a8428e webrtcsink: dump discovery pipelines on state changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1238>
 2023-06-08 15:32:49 +02:00
Mathieu Duponchelle 7447d95f1b webrtc/signalling: fix race condition in message ordering 
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>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle de0f7a08fe gstwebrtc-api: fix firefox errors about more than two stun servers 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 13:24:45 +02:00
Mathieu Duponchelle cd4b90fef4 webrtcsink/utils: remove unused decoders field in DecodingInfo 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1236>
 2023-06-08 01:54:13 +02:00
Mathieu Duponchelle 271b583876 webrtcsink: avoid panic on unprepare from an async tokio context 
.. and log an error with advice on how to dispose of elements properly
from a tokio runtime.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1218>
 2023-06-07 19:57:19 +00:00
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
Thibault Saunier ab1ec12698 webrtcsink: Add support for pre encoded streams 
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>
 2023-06-06 15:32:40 +02:00
Thibault Saunier 059cdecf7d webrtc: Unify the Codec structure between sink and source 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Thibault Saunier cf32d9d668 webrtc: Move make_element to the utils 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Thibault Saunier ce42723ad2 webrtc: Minor documentation enhancement 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1194>
 2023-06-06 15:31:45 +02:00
Mathieu Duponchelle 6346d5608e net/aws/transcriber: track discont offset in input stream 
and add it up to subsequent transcripts.

This ensures synchronization is maintained even after the input stream
experiences a discontinuity and a gap in its timestamps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1230>
 2023-06-02 08:55:11 +00:00
Mathieu Duponchelle 80582923bb aws_kvs_signaller: don't force us-east-1 region 
Instead use default region provider, with a fallback to us-east-1

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1228>
 2023-05-30 16:04:27 +00:00
Edward Hervey 31b06e52ea rtpgccbwe: Improve packet handling 
Both the delay-based *and* loss-based estimates should be computed instead of
just one. This ensures faster adaptation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1179>
 2023-05-29 08:20:36 +00:00
François Laignel 4cc2498c24 webrtcsink: use spawn_blocking instead of call_async 
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>
 2023-05-26 14:23:03 +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
Sebastian Dröge a27be7d054 net: Update to AWS SDK 0.28 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1224>
 2023-05-25 13:23:49 +03:00
François Laignel e62e9f5bd4 webrtcsink: adapt commit "abort stats collection before stopping the Signaller" 
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>
 2023-05-24 21:35:39 +02:00
Sebastian Dröge e3c46b40a0 whipsink: Request pads with webrtcbin's pad templates and not our own 
It's invalid to request pads with a pad template that is not part of the
element, and results in a critical warning.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1223>
 2023-05-24 14:14:32 +00:00
Mathieu Duponchelle 44a395f134 webrtcsink: further refactor connection to stats signals 
- Stop passing webrtcbin around without using it

- Stop using glib::closure as clippy complains when using a unit type
  default-return

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1217>
 2023-05-24 13:35:26 +02:00
Mathieu Duponchelle e13124a426 webrtcsink: fix stats_sigid logic 
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>
 2023-05-24 13:35:26 +02:00
Mathieu Duponchelle ccf076ed1e webrtcsink: don't panic in twcc-stats callback 
If webrtcbin was disposed of at this point, simply return

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/345
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1217>
 2023-05-24 13:35:26 +02:00
François Laignel 9a59763df1 webrtcsink: wait for Sessions to end 
`State::finalize_session()` asynchronously sets the Session pipeline to Null.
In some cases, sessions `webrtcbin` could terminate their transition to Null
after `webrtcsink` had reached Null.

This commit adds a set of `finalizing_sessions`. When the finalization process
starts, the session is added to the set. After `webrtcbin` has reached the Null
state, the session is removed from the set and a condvar is notified.

In `unprepare`, `webrtcsink` loops until the `finalizing_sessions` set is
empty, awaiting for the condvar to be notified when it's not.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1221>
 2023-05-24 10:18:47 +02:00
François Laignel b68e2a1ed0 webrtcsink: remove unneeded mut 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1221>
 2023-05-24 10:18:43 +02:00
Thibault Saunier 04e35e86d6 webrtcsrc: Do not pass raw caps in the transceiver 
That was not making sense.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1214>
 2023-05-18 18:23:56 +03:00
Thibault Saunier e73d7082a6 webrtcsrc: Fix caps used when creating transceiver 
We used to pass all media keys and attributes to the caps which
incorrect. Instead we should be using only the keys from the map
and remove all information related to rtcp which is irrelevant
to create the transceiver.

This also simplifies the code.

New caps look like:

```
Caps(
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 96,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP8",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 102,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 104,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 106,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 108,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "constrained-baseline",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 127,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "1",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 39,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "H264",
        packetization-mode: (gchararray) "0",
        profile: (gchararray) "main",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 98,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "0",
    },
    application/x-rtp(memory:SystemMemory) {
        media: (gchararray) "video",
        payload: (gint) 100,
        clock-rate: (gint) 90000,
        encoding-name: (gchararray) "VP9",
        profile-id: (gchararray) "2",
    },
)
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1214>
 2023-05-18 18:23:56 +03:00
François Laignel 7ba0073052 use Pad builders for optional name definition 
Also, apply auto-naming in the following cases

* When building from a non wildcard-named template, the name of the template is
  automatically assigned to the Pad. User can override with a specific name by
  calling `name()` on the `PadBuilder`.
* When building with a target and no name was provided via the above, the
  GhostPad is named after the target.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/448
Auto-naming discussion: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1255#note_1891181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
 2023-05-12 12:55:31 +02:00
François Laignel 8e93d294e5 Update to argumentless {Bin,Pipeline}::new 
See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/449

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
 2023-05-12 12:55:31 +02:00
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
François Laignel 092ae1fec8 net/webrtc: sink: add signal "request-encoded-filter" 
The new "request-encoded-filter" signal is emitted when the encoder and related
elements are added to the pipeline. When defined, the element returned by the
signal is inserted between the encoder and the payloader.

The transformation can be reverted using the [insertable streams API] on the
receiver side.

[insertable streams API]: https://developer.mozilla.org/en-US/docs/Web/API/Insertable_Streams_for_MediaStreamTrack_API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
 2023-05-09 11:17:32 +02:00
François Laignel dc5ddd3022 net/webrtc: sink: abort stats collection before stopping the Signaller 
In some rare cases, the webrtc-test entered a deadlock while executing
`WebRTCSink::unprepare`. Attaching gdb to a blocked instance showed:

* `gstrswebrtc::signaller:👿:Signaller::stop()` parked, waiting for a
  `Condvar` in `Signaller::stop()`. This was most likely awaiting for the
  receive task to complete while it was locked in `element.end_session()`.
  This code path is triggered from `unprepare` with the `State` `Mutex` locked.
* `webrtcsink:👿:WebRtcSink::process_stats` waiting for a contended `Mutex`,
  which is also the `State` `Mutex`. This prevented completion of the signal
  `gst_webrtc_bin_get_stats`.

This commit aborts the task in charge of periodically collecting stats and
ensures any remaining iteration completes before requesting the Signaller to
stop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
 2023-05-09 10:26:11 +02:00
François Laignel eca269cbf2 net/webrtc: src: don't set stun-server on webrtcbin when our property is None 
... otherwise an error occurs about the stun-server address being an empty
string which doesn't comply with the expected address format.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1202>
 2023-05-09 10:26:07 +02:00
Sebastian Dröge cb5b527d74 Update to AWS SDK 0.27 and async-tungstenite 0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1199>
 2023-05-02 15:30:00 +03:00
Sebastian Dröge 5451035215 Update async-tungstenite and AWS SDK dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1187>
 2023-04-21 10:48:10 +00:00
Sebastian Dröge cc3646640e Fix a couple of new Rust 1.69 clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1186>
 2023-04-20 16:47:45 +03:00
Edward Hervey 721d17e181 rtpgccbwe: Don't process empty lists 
The structure parsing could result in an empty vector. Don't do any processing
since the loss code assumes it's non-empty for average estimates which would
result in weird/invalid results.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1181>
 2023-04-15 19:35:27 +02:00
Mathieu Duponchelle dbdb9bc164 webrtcsink: fix navigation data channel 
At some point, presumably recently, the data channel stopped being
requested in Ready, making webrtcbin refuse to create it.

There was quite a lot of churn recently so I couldn't pinpoint the
breaking commit easily.

Fix by simply restoring the correct behavior of requesting the channel
after going to the Ready state

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/341

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1180>
 2023-04-14 14:26:22 +02:00
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
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
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
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 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
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 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
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
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
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 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
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 2b32d00589 net/aws/transcriber: use two queues for sending transcript items 
* A queue dedicated to transcript items not intended for translation.
* A queue dedicated to transcript items intended for translation. The items are
  enqueued after a separator is detected or translate-lookahead was reached.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
 2023-03-16 20:29:31 +01:00
François Laignel 5a5ca76d9d net/aws/transcriber: desambiguify SrcPad output items queue 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1137>
 2023-03-16 12:41:07 +01:00
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 3b3f0c1a29 net/aws/transcriber: fix transcript-lookahead prop nick 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1136>
 2023-03-14 21:11:33 +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
Sebastian Dröge 4eccd30ce2 Revert "aws: Temporarily enable the default features of the test-with crate" 
This reverts commit 42116b5bce.
 2023-03-14 13:28:28 +02:00
Sebastian Dröge 42116b5bce aws: Temporarily enable the default features of the test-with crate 
Version 0.9.4 fails compiling without them enabled.

See https://github.com/yanganto/test-with/pull/57
 2023-03-14 09:19:26 +02:00
Sebastian Dröge c1bac30694 webrtc: Update to aws 0.54/0.24 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1131>
 2023-03-11 09:37:14 +02:00
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
Sebastian Dröge fc5ed15af5 Update for gst::Element::link_many() and related API generalization 
Specifically, get rid of now unneeded `&`.
 2023-03-09 16:46:52 +02:00
François Laignel b9cd71d8eb net/aws/transcriber: fix eos not being sent 
For eos to be sent from the srcpad task loop, we need to go through `dequeue`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1122>
 2023-03-09 13:07:03 +01:00
François Laignel 2ea9f147ab net/aws/transcriber: fix deadlock when the pipeline is interrupted 
... also makes sure to abort the taks_iter Future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1122>
 2023-03-09 13:07:03 +01:00
Sebastian Dröge 3ef8a48ded Fix a few new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1120>
 2023-03-07 08:47:01 +00:00
Vivia Nikolaidou cd74d01324 ndisinkcombiner: Properly handle caps changes 
We are caching one video buffer, so previously we were changing the src
caps one buffer too early.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1110>
 2023-03-01 12:30:54 +00:00
François Laignel 4a988aaeb8 net/aws/transcriber: use a TranscriberLoop struct 
This helps gather together the details related to the `TranscriberLoop`.
One difference with previous implementation is that the ws `Client` is
build each time the loop is started instead of being reused. With the new
approach, we don't keep the connection open after EOS and we should be
more resistant in case of a connection failure.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
 2023-03-01 08:47:58 +00:00
François Laignel f1a080c94e net/aws/transcriber: own transcription items 
So that we can avoid copying the content.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
 2023-03-01 08:47:58 +00:00
François Laignel 36ae29d746 net/aws: enqueue transcribed buffers within the ws loop 
Instead of sending transcription events to the src pad loop, this commit
enqueues the transcribed buffers immediately in the ws loop, then notifies
the src pad loop. The src pad loop is only in charge of dequeuing the buffers.

This should help with upcoming evolutions.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1104>
 2023-03-01 08:47:58 +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
François Laignel 57f365979c net/aws: remove aws_ from the aws_transcribe* folder names 
Those folders reside under `aws`, so there's shouldn't be any confusion.

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
Thibault Saunier 0ae637f531 webrtcsink: Move RUNTIME to the crate so it can be reused 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 17:57:14 -03:00
Thibault Saunier 4ec441560b webrtc: Enhance debug messages when using unknown peer ID 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/932>
 2023-02-28 19:28:51 +00:00
Matthew Waters 542c7e12b8 webrtcsink: also support nvvidconv in lieu of nvvideoconvert 
nvvideoconvert may not exist and nvvidconv might on some Jetson
platforms.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1107>
 2023-02-28 10:12:36 +11:00
Sebastian Dröge 9fc1404415 Update minimum supported Rust version to 1.66 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
 2023-02-20 11:09:01 +02:00
Arun Raghavan 487d7fb26b hlssink3: Allow GIOStream signal handlers to return None 
If creating a playlist or fragment stream fails (disk is full, the
directory is removed, ...), we will currently crash because the signal
handler expects a non-None GIOStream. The actual callback is allowed to
return None values and we handle this in the caller, so let's not have
this restriction on the signal handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1093>
 2023-02-14 11:25:44 -05:00
Sebastian Dröge 04e101c605 Optimize various error message / debug message formatting 
Directly make use of format strings instead of formatting a string
beforehand and then passing it to the macros.
 2023-02-13 11:50:57 +02:00
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
Seungha Yang 6420fe43da rtpav1pay: Fix Leb128Bytes size parsing 
There are multiple ways of encoding the value, and don't assume
that bitstream used the way used in this plugin. Instead, count
the number of used bytes.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/312
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1090>
 2023-02-10 18:47:52 +00:00
Sebastian Dröge ac8afc4ac0 Update to async-tungstenite 0.20 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1087>
 2023-02-10 13:03:07 +02:00
Sebastian Dröge 1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +02:00
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
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 0ed74d0aa4 rtpgccbwe: Don't use clamp() if there's no clear min/max value 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/305

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1078>
 2023-02-06 21:56:46 +02:00
Sanchayan Maity 6006a0ba36 aws/s3hlssink: Fix deadlock on EOS 
In state change to NULL, we take state lock and call stop. When stop
is called, we will try to upload queued segments in S3 request thread.
That tries to take the state lock again and deadlocks.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1076>
 2023-02-03 19:09:18 +05:30
Sanchayan Maity 41aa1e51da aws/s3hlssink: Use factory name when checking name of child element 
Commit ad3f1cf fixed the name of hlssink child element to be the same
for hlssink2 and hlssink3. However, we rely on element name to return
boolean in case of hlssink3 or None in case of hlssink2 as the return
value of the delete-fragment closure.

Fix this by using the factory name instead of the element name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1076>
 2023-02-03 19:08:40 +05:30
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 194c4e9e9f rtpav1pay: Consider the marker flag to output packets immediately at the end of a frame 
Otherwise it is necessary to wait for the beginning of the following
frame, which unnecessarily increases the latency.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/255

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge 49350f738f rtpav1depay: Fix depayloading of packets starting with a leading OBU fragment followed by more OBUs 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/288

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge 1756d7a516 rtpav1depay: Fix error handling 
Don't error out immediately on errors anymore but try again with the
next packet.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/289

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge ed4e9a50d5 rtpav1depay: Set DISCONT flag on buffers following a corrupted packet 
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 560bdc4cb7 Update for glib API changes 2023-01-31 12:24:07 +02:00
Sebastian Dröge a1cce9b796 aws: Update to AWS SDK 0.54/0.24 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1066>
 2023-01-27 22:10:23 +02:00
Sebastian Dröge 3b4c48d9f5 Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1062>
 2023-01-25 10:31:19 +02:00
Arun Raghavan ad3f1cf534 aws: s3hlssink: Fix the name of the hlssink child element 
It's easier to set child element properties if the name doesn't depend
on the factory.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1061>
 2023-01-24 18:56:46 +00:00
Sebastian Dröge 2c386fb792 Update for various deprecated APIs 2023-01-22 20:07:26 +02:00
Sebastian Dröge 4582ae91ab Move remaining plugins to ParamSpec builders 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1054>
 2023-01-21 18:34:55 +02:00
Sebastian Dröge 458b2386ed Update for glib API changes 2023-01-21 18:13:48 +02:00
Sebastian Dröge 7cfd570c15 onvif: Update for allocation query caps API changes 2023-01-19 16:38:06 +02:00
Sebastian Dröge 812df78b75 webrtcbin: Update for StreamProducer API changes 2023-01-16 16:36:41 +02:00
Sebastian Dröge 6132788b02 Update for caps/structure-related string API changes 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1048>
 2023-01-15 22:58:44 +02:00
Sebastian Dröge 0c954135a3 aws: Update to AWS SDK 0.53/0.23 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1047>
 2023-01-14 18:58:30 +02:00
Mathieu Duponchelle 1a8abde884 webrtcsink: fix panic on pre-bwe request error 
We dispose of consumer pipelines asynchronously, potentially after the
session objects have been disposed of.

As session objects are the owner of the cc element, it is entirely
possible for the bwe-request signal to get emitted after cc has been
disposed of, as the closure only takes a weak reference to it.

Fix by simply checking if cc is None

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1044>
 2023-01-11 15:09:45 +00:00
Sebastian Dröge be72fefb18 reqwest: Update for API changes 2023-01-06 12:52:30 +02:00
Sebastian Dröge 781fd1df9a aws: Update to test-with 0.9 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1035>
 2023-01-05 12:35:42 +02:00
Sebastian Dröge 27435ad82e Update for API changes 2023-01-05 12:33:54 +02:00
rajneeshksoni d846f527af awss3hlssink: Add stats property. 
application can monitor the progress of hls segment generation
and upload progress.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1022>
 2023-01-04 12:36:13 +00:00
Philippe Normand 0fd63ece7d rtpav1depay: Implement srcpad set_caps 
Without this auto-pluggers such as decodebin or parsebin will be unable to
process AV1 RTP payloads.

Tested with: `videotestsrc num-buffers=50 ! videoconvert ! av1enc ! av1parse ! rtpav1pay ! queue ! decodebin3 ! videoconvert ! queue ! autovideosink`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1034>
 2023-01-03 19:35:45 +02:00
Zhao, Gang 9fa838e366 webrtc: Fix rustfmt errors 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-27 11:12:54 +02:00
Zhao, Gang 877a9bd7f3 webrtc: Share runtime between webrtcsink and signaller crates 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Zhao, Gang 1ffeb4d44d webrtc: Move from async-std to tokio 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Zhao, Gang 2bc29c1fd3 webrtc: examples: Update package-lock.json 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1019>
 2022-12-26 23:10:40 +00:00
Sebastian Dröge 4e444a066c aws: Update to AWS SDK 0.52/0.22 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1020>
 2022-12-18 07:54:30 +00:00
Mathieu Duponchelle e5360ff431 webrtc/README: update command to run the signalling server 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/277

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1012>
 2022-12-13 12:47:26 +01:00
Sebastian Dröge 3f904553ea Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1011>
 2022-12-13 11:43:16 +02:00
Sebastian Dröge 289e8a08c3 webrtchttp: Remove unnecessary clippy warning override 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1009>
 2022-12-12 14:32:12 +02:00
Sebastian Dröge fb42cd8a0f net: Update to async-tungstenite 0.19 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1005>
 2022-12-11 12:54:24 +02:00
Sebastian Dröge 9b964db4c9 whipsink: Handle offer creation errors more gracefully 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 12:15:55 +02:00
Sebastian Dröge 8452cd9efa webrtchttp: Fix missing import for docs build 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 12:10:53 +02:00
Sebastian Dröge 9c31344bbc webrtchttp: Don't use let-else for now 
We still support Rust 1.63.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 12:08:57 +02:00
Sebastian Dröge 5dc52975ff webrtchttp: Fix formatting 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 12:07:09 +02:00
Sanchayan Maity 40680a47ab webrtchttp: Use tokio runtime for spawning thread used for candidate offer 
While at it, we had a bug in whepsrc where for redirect we were
incorrectly calling initial_post_request instead of do_post. Fix
that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 12:27:07 +05:30
Sanchayan Maity d18761892e webrtchttp: Use a proper Rust type name for ICE transport policy 
We don't need to namespace here but can just use the Rust namespaces.
Only the GType name has to stay like it is.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity 2eba3b321e webrtchttp: Do not import element_imp_error 
element_imp_error and such macros should not be imported but rather
only be accessed via gst namespace.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity 0b1b8b91b9 webrtchttp: Do not block webrtcbin signal handlers for sending candidates 
While at it, drop the OPTIONS request in WHIP sink. This was not really
required. See section 4.4 of the spec
https://www.ietf.org/archive/id/draft-ietf-wish-whip-01.html#name-stun-turn-server-configurat

Also introduce a new error type and distinguish between a future being
aborted or returning an error.

We call abort only during shutdown and hence except for the DELETE
resource request being aborted, other waits on future should not
be fatal.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Alba Mendez db39370701 webrtchttp: whipsink: construct TURN URL correctly 
Right now the code manually pieces together the components
in a String for efficiency. When credentials contain special
characters this can result in invalid URLs, so do it the proper
way (with Url::parse + format) to make sure components are escaped
as needed.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity 9fb058d5bc webrtchttp: Drop unused dependencies 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity b5daa92c9d webrtchttp: Implement timeout for waiting on futures 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity cc7419308b webrtchttp: whipsink: Add candidates when sending the offer 
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>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity b992596236 webrtchttp: whipsink: Miscellaneous clean up 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity b427cb6a3d webrtchttp: Factor out the common bits for WHIP and WHEP 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/949>
 2022-12-05 11:04:45 +05:30
Sanchayan Maity 6be5796888 Add a WebRTC WHEP source element 
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>
 2022-12-05 11:04:45 +05:30
Raphael Dürscheid aa2abc50bf webrtcsink: Support nvv4l2vp9enc 
Naive support for nvv4l2vp9enc by assuming configuration is equivalent
to existing nvv4l2vp8enc. Validated to have relevant properties.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/983>
 2022-12-02 10:18:27 +00:00
Jordan Petridis 821c23e202 net/ndi: fix build with --no-default-features 
doc_show_default() is only available with gst/v1_18

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/588>
 2022-11-29 21:06:12 +02:00
Vivia Nikolaidou 5bbe0eab25 ndisrc: Use actual number of channels in positions_from_mask 
Otherwise it fails for mono and stereo

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/991>
 2022-11-29 12:19:45 +02:00
Vivia Nikolaidou 73ce616bd9 ndisrc: Use default channel mask for audio output 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/988>
 2022-11-28 17:06:07 +02:00
Sebastian Dröge fceacf7081 Update for gst::Array / gst::List API improvements 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/985>
 2022-11-27 01:12:46 +02:00
Sebastian Dröge 0e2a00cbc8 aws: Update to env_logger 0.10 for the tests 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/984>
 2022-11-25 11:08:19 +02:00
Sebastian Dröge 456fb276d6 Revert "Update for pango API changes" 
This reverts commit 6e54d3cea9.

The change was wrong and the pango bindings work the same as before
again.
 2022-11-18 10:58:41 +02:00
Sebastian Dröge 6e54d3cea9 Update for pango API changes 
pango::Language::from_string() can fail and also can accept None as
argument.
 2022-11-18 09:46:50 +02:00
Thibault Saunier 6b11284e8a webrtcsink: Make the turn-server prop a turn-servers list 
So that we can simply specify several turn servers at once

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/973>
 2022-11-16 14:48:16 +00:00
Arun Raghavan 3abd13e57b aws: s3sink: Treat stopping without EOS as an error for multipart upload 
This allows us to try to clean up based on configuration (abort /
complete / do nothing) if the pipeline is shut down without an EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/970>
 2022-11-15 02:28:35 +00:00
Guillaume Desmottes 37cb636140 webrtc: README: fix couple of links 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/975>
 2022-11-11 14:51:46 +01:00
Mathieu Duponchelle 66e7b314f7 webrtcsink: improve debug 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/972>
 2022-11-10 15:00:19 +00:00
Sebastian Dröge a5f3197651 Add missing doc features to WebRTC plugins 2022-11-07 18:06:29 +00:00
Jan Beich 9aeaac5a96 ndi: provide Unix fallback after 3fe9e4a207 
error[E0425]: cannot find value `LIBRARY_NAME` in this scope
   --> net/ndi/src/ndisys.rs:336:23
    |
336 |             path.push(LIBRARY_NAME);
    |                       ^^^^^^^^^^^^ not found in this scope

error[E0425]: cannot find value `LIBRARY_NAME` in this scope
   --> net/ndi/src/ndisys.rs:339:33
    |
339 |             path::PathBuf::from(LIBRARY_NAME)
    |                                 ^^^^^^^^^^^^ not found in this scope
 2022-11-05 02:51:28 +00:00
Arun Raghavan 54c84a7211 aws: Skip s3 test on Windows until we figure out why it times out 2022-11-02 13:14:08 -04:00
Sebastian Dröge a8250abbf1 Fix various new clippy warnings 2022-11-01 10:27:48 +02:00
Sebastian Dröge 976ae5707e webrtc: Update to human_bytes 0.4 2022-10-31 14:11:29 +02:00
Sebastian Dröge 6ceeadc0f0 aws: Update to aws 0.21/0.51 2022-10-31 14:11:29 +02:00
Sebastian Dröge ce166b4d8f whipsink: Add object to debug logs 2022-10-26 16:20:26 +03:00
Guillaume Desmottes d46857d3b1 aws: fix title in README 
The title was not matching the actual plugin name which was confusing.
 2022-10-26 11:13:47 +02:00
Sebastian Dröge bf6bdab80c webrtc: Remove version requirement from internal crate dependencies 2022-10-24 19:50:24 +03:00
Sebastian Dröge f2223cf2cb Update versions to 0.10.0-alpha.1 2022-10-24 19:31:19 +03:00
Sebastian Dröge b64f951160 Update to async-tungstenite 0.18 2022-10-24 18:03:33 +03:00
Sebastian Dröge 9a68f6e221 Move from imp.instance() to imp.obj() 
It's doing the same thing and is shorter.
 2022-10-23 23:08:46 +03:00
François Laignel 86776be58c Remove & for obj in log macros 
This is no longer necessary.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1137
 2022-10-23 21:22:31 +02:00
Sebastian Dröge f045099fc1 Fix GObject type names, GStreamer debug category names and element factory names 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/198
 2022-10-23 20:46:08 +03:00
Sebastian Dröge 5d44e0eb3c rtp: Move GCC bandwidth estimation element from webrtc to rtp plugin 2022-10-23 20:25:08 +03:00
Sebastian Dröge 20ad9175d8 Make GStreamer plugin/crate/library/directory names and descriptions consistent 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/238
 2022-10-23 20:25:08 +03:00
Sebastian Dröge 45168639e9 Rename rtpav1 plugin to just rtp 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/243
 2022-10-23 20:04:43 +03:00
Sebastian Dröge f058a5e229 Various minor cleanups 2022-10-22 19:50:24 +03:00
François Laignel 6319d104a8 Take advantage of Into<Option<_>> args 
Commit 24b7cfc8 applied changes related to nullability as declared
by gir. One consequence was that some functions signature ended up
requiring users to pass `Some(val)` when they could use `val`
before.

This commit applies changes on `gstreamer-rs` which, will honoring
the nullability stil allow users to pass `val` for the few affected
functions.

This commit also fixes the signature for `Element::request_new_pad`
which was updated upstream.
 2022-10-21 11:54:24 +02:00
Sebastian Dröge 7b5d887c5b onvifmetadatacombiner: On timeout don't wait for metadata to arrive anymore but output the current video frame 
Otherwise it will be too late downstream.
 2022-10-21 07:08:46 +00:00
Sebastian Dröge 09ffeaf04e onvifmetadatacombiner: Add a lot of trace debug output 2022-10-21 07:08:46 +00:00
Thibault Saunier 5c89c3db69 webrtc: Rename and add to meson build the signalling server 
The binary was only called `server` it has been renamed to
`gst-webrtc-signalling-server` and is installed in meson.
 2022-10-20 18:20:15 +00:00
Thibault Saunier cbdd3a7f26 webrtc: Enhance documentation 2022-10-20 12:04:43 +00:00
Sebastian Dröge c0bf05d4bb webrtc: Minor cleanup 2022-10-20 13:20:32 +03:00
Thibault Saunier 71ed04d89b webrtc: Rename signaller and protocol crates 2022-10-20 13:32:31 +02:00
Thibault Saunier 25bda89ac8 webrtc: Update an unify rust-version and edition 
So it all matches the rest of the plugins
 2022-10-20 13:32:31 +02:00
Thibault Saunier 4942a916a8 webrtc: Uniformise GType names 2022-10-20 13:32:31 +02:00
Thibault Saunier 37c0239aff webrtc: Port to new ElementBuilder API 2022-10-20 13:32:31 +02:00
Thibault Saunier ad78936365 webrtc: Enable more documentation 2022-10-20 13:32:31 +02:00
Thibault Saunier 0f0dec7fa9 webrtc: Fix fmt issues 2022-10-20 11:51:59 +02:00
Thibault Saunier 5ab7be6124 webrtc: Add SDPX license header on every file 2022-10-20 11:51:58 +02:00
Thibault Saunier 39c0dcb0d4 Plug webrtc in 2022-10-20 11:51:58 +02:00
Thibault Saunier b164daf510 webrtc: Fix clippy issues 2022-10-20 11:51:58 +02:00
Thibault Saunier 87fd49a9bf webrtc:signalling: Remove short option for 'host' in the cli 
It clashes with `--help`
 2022-10-20 11:51:58 +02:00
Thibault Saunier eb9d0bb824 Merge 'webrtcsink' from 020c7e2900 2022-10-20 11:51:58 +02:00
Sebastian Dröge 12400b6b87 Update everything for element factory builder API changes 
And set properties as part of object construction wherever it makes
sense.
 2022-10-19 19:43:29 +03:00
Sebastian Dröge 9ce8e93c63 rtpav1pay: Track last known upstream PTS/DTS in case not all OBUs are properly timestamped 2022-10-19 15:42:48 +03:00
Sebastian Dröge 36861edf9a rtpav1pay: Use a VecDeque instead of a Vec for the queued OBUs 
And use a `Vec` plus offset for consuming partial byte buffers.
Removing the first element from a `Vec` repeatedly is not very cheap.

Also simplify calculation of the current packet by removing a mostly
unused type and keeping track of the calculations always locally instead
of sometimes storing it in the element state.
 2022-10-19 15:23:10 +03:00
Sebastian Dröge 24b7cfc841 Update for GStreamer API changes 2022-10-18 19:26:52 +03:00
Arun Raghavan 03b03fe2dd whipsink: Log error body along with status code when POST fails 2022-10-18 17:01:36 +02:00
Thibault Saunier 5e7537953c webrtc: Move to net/webrtc 2022-10-18 15:18:53 +02:00
Sanchayan Maity c63307e6d7 net/webrtc-http: whipsink: Return a proper error message & not panic 
On a server error, we currently crash and panic. Return a proper error
message instead.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/914>
 2022-10-18 10:38:57 +00:00
François Laignel 8011eadfd2 Use new format constructors 
See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1128
 2022-10-18 10:36:59 +00:00
Arun Raghavan e66378d254 aws: Add a test for s3src/s3sink 
This does rely on AWS credentials being provided in the environment, but
the test will be ignored if those are missing.
 2022-10-18 09:51:34 +00:00
Sebastian Dröge e17688a2da Update for pango API changes 2022-10-17 20:02:02 +03:00
Vivia Nikolaidou 0ab965335f onvifmetadataoverlay, cea608overlay: Fix pangocairo::FontMap::new() 
It doesn't return an Option anymore.
 2022-10-14 18:12:33 +03:00
Vivia Nikolaidou f11b0fa5eb plugins, examples, tutorials: Use AudioCapsBuilder and VideoCapsBuilder 
Simplify caps creation code
 2022-10-13 19:24:57 +00:00
Sebastian Dröge 862c2af1d9 ndi: Remove unnecessary explicit Send+Sync impls 
These are automatically available now.
 2022-10-13 17:54:08 +00:00
Vivia Nikolaidou dbd5a44b90 hlssink3: Use #[cfg(feature = "doc")] on gst::prelude import 
It otherwise gives a warning about the unused import
 2022-10-13 14:22:36 +03:00
Sebastian Dröge 5f19639d0f ndi: Various code cleanup 2022-10-13 08:52:52 +00:00
Sebastian Dröge b2ddb34258 onvif: Switch from minidom to xmltree for parsing ONVIF timed metadata 
minidom doesn't handle various valid but suboptimal XML documents.
 2022-10-12 21:00:13 +00:00
Sebastian Dröge 97e0852156 ndi: Add NDI plugin to the docs 2022-10-12 22:25:13 +03:00
Sebastian Dröge 53b02a82ae ndi: Re-organize code a bit and don't make internal modules public 2022-10-12 22:09:56 +03:00
Sebastian Dröge 0a2e6e47c9 ndi: Silence some more clippy warnings 2022-10-12 22:09:55 +03:00
Sebastian Dröge db8037d16c ndi: Update for pad default functions API changes 2022-10-12 22:09:55 +03:00
Sebastian Dröge 3fe9e4a207 ndi: Implement dynamic loading of the NDI SDK 
And build the plugin on the CI and via meson.
 2022-10-12 22:09:53 +03:00
Sebastian Dröge 16c036e2cc ndi: Make element factory details and debug categories more consistent 2022-10-12 21:29:07 +03:00
Sebastian Dröge 907910329f ndi: Prefix GType names with Gst 2022-10-12 21:29:07 +03:00
Sebastian Dröge 047f990c78 ndi: Integrate into the build system 2022-10-12 21:29:07 +03:00
Sebastian Dröge a000432b13 ndi: Relicense plugin from LGPL-2.1 to MPL-2 
This was agreed to by all previous contributors in writing.
 2022-10-12 21:29:07 +03:00
Sebastian Dröge fb8192f40b ndi: Remove unnecessary reference-timestamps feature 2022-10-12 21:29:07 +03:00
Vivia Nikolaidou fedd67dcaa ndi: Use AudioCapsBuilder and VideoCapsBuilder 
Simplify caps creation codes
 2022-10-12 21:29:07 +03:00
Vivia Nikolaidou 95e8deded9 ndi: Simplify code using ParamSpecBuilder 2022-10-12 21:29:07 +03:00
Vivia Nikolaidou 77a5e35081 ndi: Update to git version of the bindings 2022-10-12 21:29:07 +03:00
Vivia Nikolaidou 18cbb587ba ndisrcdemux: Add no-more-pads signal 
Emit no-more-pads if we are adding the second pad of the element.
 2022-10-12 21:29:07 +03:00
Sebastian Dröge 1c43a51520 ndisrcdemux: Use ANY caps in the pad templates of ndisrcdemux 
When using the Advanced SDK it is possible to output compressed formats
too.
 2022-10-12 21:29:07 +03:00
Sebastian Dröge 26f843a89f ndisrc: Fix latency reporting in auto timestamp mode 2022-10-12 21:29:07 +03:00
Sebastian Dröge 9c10ba87df ndisrc: Improve handling of broken sources with regards to timestamping 
- NDI HX Camera Android in the past used 1ns instead of 100ns as unit
   for timecodes/timestamps.
 - NDI HX Camera iOS uses 0 for all timecodes and the same non-zero
   value for all audio timestamps

Detect such situations and try to compensate for them. Also add a new
"auto" timestamping mode that prefers to use timecodes and otherwise
falls back to timestamps or receive times.

Fixes https://github.com/teltek/gst-plugin-ndi/issues/79
 2022-10-12 21:29:07 +03:00
Sebastian Dröge a3c752830b ndisrc: Keep track of audio/video and timestamp/timecode observations separately 
Audio/video are in practice not always from the same clock and can have
different behaviours with regards to clock rate and jitter. Handling
them separately generally gives better results for the timestamps output
by the source element.
 2022-10-12 21:29:07 +03:00
Sebastian Dröge b82acb9ca9 ndisrc: Remove unnecessary Arc around the timestamp observations and use AtomicRefCell instead of Mutex 2022-10-12 21:29:07 +03:00
Sebastian Dröge 718734ab18 ndi: Fix/silence various clippy warnings 2022-10-12 21:29:07 +03:00
Sebastian Dröge 7a90500fe7 Merge branch 'master' of https://github.com/teltek/gst-plugin-ndi 2022-10-12 21:27:56 +03:00
Sebastian Dröge e49138516c Update for pad default functions API changes 2022-10-12 19:50:15 +03:00
Sebastian Dröge 9c540d8abb Move everything to net/ndi for preparing to merge into gst-plugins-rs 2022-10-12 19:25:32 +03:00
François Laignel bc5b51687d fix formatted values constructors 
In restrospect, building formatted values using operations on the
`ONE` constant doesn't seem idiomatic. This commit uses new panicking
constructors instead.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1122
 2022-10-11 15:06:53 +02:00
François Laignel bd14e476f1 Fix direct access to the inner specific formatted values 
This is no longer available as this could lead to building a defined
value in Rust which could be interpreted as undefined in C due to
the sentinel `u64::MAX` for `None`.

Use the constants (e.g. `ONE`, `K`, `M`, ...) and operations to build
a value and deref (`*`) to get the quantity as an integer.
 2022-10-10 19:28:13 +02:00
Sebastian Dröge 7ee4afacf4 Change *Impl trait methods to only take &self and not Self::Type in addition 2022-10-10 15:03:25 +03:00
Sebastian Dröge 4c57a97d4d Update for glib::Object::new() API changes 2022-10-07 23:54:53 +03:00
Nirbheek Chauhan 1d4d3e4cb0 build: Update versions to be 0.9.0-alpha.1 
0.9.0 is the next release, so we can't name things that already.

Also the version in meson.build was 0.13.0, which is completely wrong.
 2022-10-04 21:27:23 +05:30
Sebastian Dröge 8601562efe onvif: Fix for gst::meta::CustomMeta::register() API change 2022-09-29 17:48:27 +03:00
Sebastian Dröge 0b81ed2e34 rtpav1: Use GStreamer types by namespace instead of importing dozens of types directly into the scope 
For consistency with other plugins and to reduce confusion of where
types actually come from.
 2022-09-28 08:14:07 +00:00
Sebastian Dröge 5774d9c9ee rtpav1: Reset state on FlushStop/Eos in all conditions and reset all of the state 2022-09-28 08:14:07 +00:00
Sebastian Dröge d6ab55c263 onvifmetadataparse: Schedule EOS events after the last currently queued up frame 
Otherwise EOS might be sent before the last frame's data, or even at a
much earlier frame due to reordering.
 2022-09-27 11:43:54 +00:00
Sebastian Dröge f0b2df49dc onvifmetadataparse: Handle negative running times in debug output 2022-09-27 11:43:54 +00:00
Sebastian Dröge 692a063528 onvifmetadataparse: Refactor clock/condvar waiting 
Always first try draining queued data in the loop and only start waiting
if there's nothing to drain right now. Otherwise data might have to be
drained right now but we still wait and nothing is ever waking up the
source pad task again.

Also make sure to not wait multiple times on the same gst::ClockId but
instead unset it after waiting on it and no new one was scheduled in the
meantime. Future waits on the same ClockId will immediately return and
instead we should wait on the condvar if no new ClockId is available.
 2022-09-23 13:26:15 +03:00
Sebastian Dröge c4d2f4a60a onvifmetadataparse: Start source pad task on StreamStart if needed 
Otherwise receiving StreamStart after Eos might keep the source pad task
paused and no new data is ever pushed downstream.
 2022-09-23 13:26:15 +03:00
François Laignel 0b7259afac Fixes for removal of SpecificFormattedValues ops on ref 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/874>
 2022-09-22 12:18:49 +02:00
François Laignel caefa6d33e net/onvif: update with new gst::Signed features 2022-09-21 17:45:22 +00:00
Sebastian Dröge c32f0ca12e rtpav1: Specify version helper dependency by path 
It's in the same repository after all.
 2022-09-21 11:17:44 +03:00
Sebastian Dröge 6a10728d94 aws: Update to aws 0.48/0.18 2022-09-21 11:17:44 +03:00
Sebastian Dröge 1fa39d0ab4 onvifmetadatacombiner: Drop gap metadata buffers 
They won't have a reference timestamp metadata set and are not useful
for further processing.
 2022-09-16 14:54:33 +03:00
Sebastian Dröge f2893aae0b onvifmetadataparse: Simplify some code 2022-09-16 14:54:33 +03:00
Sebastian Dröge 49602e1e01 onvifmetadataparse: Drop initial buffers until an UTC/running time mapping can be established 2022-09-16 14:54:33 +03:00
Sebastian Dröge c6d8fec18f onvifmetadataparse: Drop initial buffers if their UTC time would be negative 2022-09-16 14:54:33 +03:00
Sebastian Dröge 28151f2011 onvifmetadataparse: Push buffers from a separate source pad task to guarantee latency and generally improve correctness 2022-09-16 14:54:33 +03:00
Sebastian Dröge 18f3edd3ee Add missing Since markers to new plugins 2022-09-15 09:40:53 +03:00
rajneeshksoni 45962eca1c s3sink, s3src: Max 1 (re)try when retry-duration < request_timeout. 
When retry-duration is less than request_timeout, only 1 try
is attempted.
 2022-09-13 08:02:54 +00:00
rajneeshksoni 62f76e1e8b s3sink: Dont set call_timeout,call_attempt_timeout is enough with retry. 
When call_timeout is triggered, request will fail
irrespective of the retry setting. call_timeout define
max time request can take along with retry.
It can be solved by either setting call_timeout to
retry * call_attempt_timeout or not setting the call_timeout.

As per thread call_attempt and rety setting is enough.
https://github.com/awslabs/aws-sdk-rust/issues/558
 2022-09-13 08:02:54 +00:00
Sebastian Dröge cc0ef5290f rtpav1depay: Don't unnecessary map RTP payload a second time 
`RTPBuffer` already has it mapped internally and can give direct access
to it as byte slice.
 2022-09-12 18:14:39 +03:00
Sebastian Dröge 7edc9e656f rtpav1pay: Don't push buffers downstream while holding mutexes 
And also push all packets that can be generated as a time as a single
buffer list instead of one by one.
 2022-09-12 18:14:39 +03:00
Sebastian Dröge f9a8e121e1 rtpav1: Remove some unneeded lifetime annotations 2022-09-12 18:14:39 +03:00
Vivienne Watermeier 8d73b5008a Add RTP de/payloader elements for AV1 
Fixes https://gitlab.freedesktop.org/gstreamer/gstreamer/-/issues/881
 2022-09-12 18:14:39 +03:00
Thibault Saunier 528bbcf67e onvifmetadatacombiner: Do not classify as Muxer 
It confuses `encodebin` and technically it is not really a muxer so
as agreed on IRC, I am proposing to remove that classification.
 2022-09-09 10:01:12 +03:00
Mathieu Duponchelle 419cc03133 awstranscriber: only set vocabulary filter when vocabulary is set 
AWS otherwise refuses to start the transcription.
 2022-09-09 06:53:54 +00:00
Mathieu Duponchelle 72b659b3ea awstranscriber: fix set_property for language-code 2022-09-09 06:53:54 +00:00
Sebastian Dröge 1a40186485 Update for GLib ParamSpec builder API changes 2022-09-05 11:45:47 +03:00
Sebastian Dröge 46dddaf31c Update minimum supported Rust version to 1.63 2022-09-04 21:31:55 +03:00
Xavier Claessens 16f9c37c71 Fix missing pkgconfig requires 2022-09-02 22:00:57 +00:00
Taruntej Kanakamalla 67e9ba8286 whipsink: A GstBin implementation for WHIP 
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
 2022-09-03 00:18:59 +03:00
Sebastian Dröge 827099d22d aws: Update to aws 0.18/0.48 2022-09-02 10:46:02 +03:00
Sebastian Dröge cb339c1bf8 onvifmetadataparse: Pass through other XML as is with the UTC times based on the buffer PTSs 2022-08-31 10:33:16 +00:00
Sebastian Dröge 420f36251a onvif: Rename onvif(de)pay to rtponvifmetadata(de)pay and include the metadata specifier in the other element names too 
This is more descriptive and avoids any future conflicts with other
kinds of ONVIF specific RTP (de)payloaders.
 2022-08-31 13:00:53 +03:00
Thibault Saunier 16d804e761 doc: Mark request::user-agent as doc show default 2022-08-29 18:33:22 -04:00
Thibault Saunier 67e651f57c Allow "unused_doc_comments" as we use hotdoc and not rustdoc 2022-08-29 18:33:22 -04:00
Thibault Saunier 31a53bba8a Generate plugins documentation using hotdoc 
Which will automatically be integrated in gstreamer documentation
 2022-08-29 18:33:22 -04:00
Mathieu Duponchelle 052092cd2e onvifmetadata: removing encoding field 
The encoding of ONVIF metadata is always UTF-8. ONVIF metadata may
or may not be encoded with gzip, but we don't see a use case for
transporting compressed ONVIF metadata between elements for now.
 2022-08-24 08:57:12 +00:00
Arun Raghavan 56e7a2f6ab aws: Document the s3hlssink element in README 2022-08-23 06:19:39 -04:00
Vivia Nikolaidou 5606111345 plugins: Simplify code using ParamSpecBuilder 2022-08-22 17:58:43 +03:00
Sebastian Dröge 374bb8323f Fix build after glib SignalBuilder::param_types() API change 2022-08-17 23:37:39 +03:00
Sebastian Dröge 9827406113 onvifmetadataparse: Use NTP reference timestamp meta 
The times are in the NTP epoch.
 2022-08-16 15:51:32 +03:00
Sebastian Dröge be56991b73 onvifmetadataparse: use NTP epoch everywhere instead of mixing UNIX/NTP epochs 2022-08-16 14:14:24 +03:00
Mathieu Duponchelle 3011764da1 onvifaggregator: refactor, expect parsed metadata 
The aggregator was consuming meta buffers too greedily, causing
potential interleaving starvation upstream. Refactor to consume
media and meta buffers synchronously

Also expect parsed=true metadata caps (requiring an upstream
onvifmetadataparse element).
 2022-08-16 12:28:52 +03:00
Sebastian Dröge 837126be76 onvifmetadataparse: Only define the namespace prefix once for the top-level element 2022-08-12 22:35:40 +03:00