Commit graph

3156 commits

Author SHA1 Message Date
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
Sebastian Dröge
b7e6e5cbc9 Update CHANGELOG.md for 0.10.8 2023-06-07 01:17:34 +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
Guillaume Desmottes
f4604e1c58 uriplaylistbin: use thiserror
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-06 12:46:17 +02:00
Guillaume Desmottes
432de060ea uriplaylistbin: example: display iterations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-05 14:09:41 +02:00
Guillaume Desmottes
97fa20237f uriplaylistbin: prevent deadlock when notifying property changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-05 14:09:41 +02:00
Guillaume Desmottes
780d9d5b78 uriplaylistbin: example display when leaving because of eos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1232>
2023-06-05 14:09:41 +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
Sebastian Dröge
2e83107c18 fmp4mux: Don't wait for more data if a stream has no GOP starting before fragment end
Simply don't output anything for this stream and only include it in the
future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1229>
2023-06-01 19:46:06 +03:00
Sebastian Dröge
a5fcd66c95 fmp4mux: Consider a stream filled if the earliest GOP starts after the current chunk
There's not going to be any buffer to output for this stream in the
current chunk.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1229>
2023-06-01 19:25:44 +03:00
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
Sebastian Dröge
dfb261ac9a Fix a couple of trivial clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1226>
2023-05-30 10:20:00 +03:00
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
Arun Raghavan
b05c21680d Revert "fmp4: Return a running time in get_next_time()"
This reverts commit 04bb7b4db0.

As Sebastian points out, the chunk PTS is already in running time, so
this was wrong from the start.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/363
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1219>
2023-05-23 09:27:00 -04:00
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
Seungha Yang
3406e604cd fallbacksrc: Don't apply fallback-audio-caps to the main audio stream
Intended behavior is configuring audio convert/resample elements
only for the fallback stream and also fallback-audio-caps is set.
Video and image stream are doing it as intended already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1213>
2023-05-17 23:49:09 +09:00
Guillaume Desmottes
7ebf2d7a4f fallbackswitch: document the pad priority ordering
I just wasted lots of time trying to figure out why my higher priority
pad wasn't used...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1208>
2023-05-15 16:13:20 +02:00
Sanchayan Maity
067d47f0ec videofx: border: Do not advertise I420 for non-zero border radius
In certain cases, roundedcorners would negotiate to I420 even when user
supplied a non-zero border radius.

For example, the below pipeline leads to I420 being negotiated even
though a non-zero border radius was given. Ideally, this pipeline
should have failed at the negotiation stage.

```bash
gst-launch-1.0 -v \
   videotestsrc num-buffers=1000 pattern=white ! \
   video/x-raw,width=320,height=180 ! \
   roundedcorners border-radius-px=10 ! videobox border-alpha=0 top=-10 left=-10 right=-10 bottom=-10 fill=yellow ! \
   compositor name=comp sink_0::xpos=960   sink_0::ypos=0  sink_0::width=320 sink_0::height=180 sink_0::alpha=1.0 sink_1::xpos=960 sink_1::ypos=180  sink_1::width=320 sink_1::height=180 sink_1::alpha=1.0  \
   sink_2::xpos=960 sink_2::ypos=360  sink_2::width=320 sink_2::height=180 sink_2::alpha=1.0 sink_3::xpos=0 sink_3::ypos=0  sink_3::width=960 sink_3::height=720 sink_3::alpha=1.0 ! \
   video/x-raw,width=1280,height=720! x264enc ! mp4mux ! filesink location=test.mp4 \
   videotestsrc num-buffers=1000 pattern=red ! \
   video/x-raw,width=320,height=180 ! roundedcorners border-radius-px=10 ! comp. \
      videotestsrc num-buffers=1000 pattern=blue ! \
   video/x-raw,width=320,height=180 ! roundedcorners border-radius-px=10 ! comp. \
      videotestsrc num-buffers=1000 pattern=green ! \
   video/x-raw,width=960,height=720 ! roundedcorners border-radius-px=10 ! comp.
```

If border radius is non-zero, we should not really allow negotiation
to select I420. Fix this by returning only A420 for border-radius > 0
in `transform_caps` instead of returning both like earlier.

Another example of a simpler pipeline like below which would earlier work

```bash
gst-launch-1.0 videotestsrc pattern=red ! videoconvert ! video/x-raw,width=1923,height=1087,format=I420 ! roundedcorners border-radius-px=40 ! video/x-raw,format=I420 ! videoconvert ! gtksink
```

now fails with

```bash
WARNING: erroneous pipeline: could not link roundedcorners0 to videoconvert1, roundedcorners0 can't handle caps video/x-raw, format=(string)I420
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1211>
2023-05-15 12:19:09 +05:30
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
Sebastian Dröge
32d59c31d8 fmp4: examples: Update to dash-mpd 0.9
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1210>
2023-05-12 09:45:40 +03:00
Seungha Yang
773fcd0780 transcriberbin: Add "language-code" property
Proxy the child transcriber element's property so that transcriberbin
can apply the property with required state management

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1205>
2023-05-10 19:12:01 +00:00
Antonio Kevo
cf21bfabf2 fmp4: Use updated start_pts when checking stream filled
After calculating the earliest pts, the fragment_start_pts and
chunk_start_pts in State are updated. However, when checking if the
stream is filled, the previous start_pts (set to None) is used instead.
This means that chunk_filled and fragment_filled will be false the first
time aggregate() is called, assuming timeout is false, all_eos is false,
and the sinkpad is not EOS. This requires aggregate() having to be
called a second time before the first fragment is sent.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1207>
2023-05-10 16:09:25 +02:00
Sebastian Dröge
8b2b12e767 Update CHANGELOG.md for 0.10.7 2023-05-09 20:48:43 +03:00
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
de1c8ece43 deny: Update 2023-05-08 18:44:12 +03:00
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
Guillaume Desmottes
76c8279101 fmp4: define minBufferTime in example mpd
Required to validate the manifest with https://beta.conformance.dashif.org/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1196>
2023-04-27 14:16:40 +02:00
Sebastian Dröge
05ee55d617 fmp4: Update example to dash-mpd 0.8
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1193>
2023-04-25 08:54:48 +03:00
Nick Steel
3d01c9b363 spotify: check cached creds username before use
If a username was specified, only use cached credentials that match
that username.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1192>
2023-04-24 15:29:22 +01:00
Jan Beich
8d6751c88d gtk4: unbreak wayland, x11egl, x11glx features on non-Linux
As the features are non-default leave the responsibility to filter by
platform to consumers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1191>
2023-04-22 12:58:29 +00:00