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Author SHA1 Message Date
Mathieu Duponchelle
9707bb89e6 webrtcsink: fix pipeline when input caps contain max-framerate 
GstVideoInfo uses max-framerate to compute its fps, but this leads
to issues in videorate when framerate is actually 0/1.

Fix this by stripping away max-framerate from input caps

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1276>
 2023-07-13 22:18:08 +02:00
Sebastian Dröge
0331522128 webrtcsink: Configure only 4 threads for x264enc 
More threads can cause more slices to be created, and Chrome simply falls
apart if there are more than a few slices and fails decoding.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1275>
 2023-07-13 16:59:43 +03:00
Sebastian Dröge
ca51cf2509 webrtcsink: Translate force-keyunit events to force-IDR action signal for NVIDIA encoders 
NVIDIA's v4l2 encoder elements don't handle the force-keyunit events but
instead provide a custom action signal based API for requesting a
keyframe.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1274>
 2023-07-12 10:09:32 +00:00
Sebastian Dröge
bbd3d9ffe0 Remove unnecessary mut everywhere 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1273>
 2023-07-11 10:09:35 +03:00
Sebastian Dröge
ee4aca3010 webrtcsink: Set config-interval=-1 and aggregate-mode=zero-latency on rtph26[45]pay 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1272>
 2023-07-10 19:48:37 +03:00
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