Commit graph

896 commits

Author SHA1 Message Date
Piotr Brzeziński
982a9a9aea hlssink3: Post hls-segment-added message
Posts a simple 'hls-segment-added' message with the segment location, start running time and duration.
With hlssink2, it was possible to catch 'splitmuxsink-fragment-closed', but since hlssink3 doesn't forward that message
(and hlscmafsink doesn't even use that mux), the new one was added to allow for listening for new fragments being added.

I extended the existing tests to check whether this message is posted correctly.
They theoretically only cover hlssink3, but hlscmafsink uses the same base class so it should be alright for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1677>
2024-08-20 18:32:59 +00:00
Sebastian Dröge
eb0a44fe67 ndisrc: Move timestamp handling from demuxer to source
This allows putting correct timestamps on buffers coming out of the
source already instead of leaving them unset until the demuxer.

And also calculate timestamps for metadata buffers.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1718>
2024-08-16 06:07:35 +00:00
Mathieu Duponchelle
1c48d7065d gstwebrtc-api example: add support for requesting mix matrix
This is one example of how a consumer might send over custom upstream
event requests to the producer.

As webrtcsink will deserialize numbers in priority as integers, we need
a custom stringifying function to ensure members of the matrix array are
indeed serialized with the floating point.

An optional stringifier parameter is thus added to the
sendControlRequest API.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 15:42:04 +00:00
Mathieu Duponchelle
01e28ddfe2 webrtcsink: implement generic data channel control mechanism ..
.. and deprecate data channel navigation in favor of it.

A new property, "enable-data-channel-control" is exposed, when set to
TRUE a control data channel is offered, over which can be sent typed
upstream events.

This means further upstream events will be usable, for now only
navigation and custom upstream events are handled.

In addition, send response messages to notify the consumer of whether
its requests have been handled.

In the future this can also be extended to allow the consumer to send
queries, or seek events ..

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1711>
2024-08-15 15:42:04 +00:00
Mathieu Duponchelle
0a6963f7ce gstwebrtc-api: example: use http by default
That way the webpage connects with ws:/ to the signaller.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/589
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1704>
2024-08-14 14:10:04 +00:00
Sebastian Dröge
102185d09d mpegtslivesrc: Handle PCR discontinuities as errors for now
More work is needed to make this work seemlessly and right now it would
simply cause invalid timestamps to be created.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1717>
2024-08-14 12:34:18 +00:00
Sebastian Dröge
ede82ca5b4 hlssink3: Don't use is-live=true
This sometimes produces imperfect timestamps that cause the fragment
duration to be slightly different than expected.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1716>
2024-08-14 13:05:40 +03:00
Sebastian Dröge
bc930122ba webrtcsrc: Make sure to always call end_session() without the state lock
This was already done in another place for the same reason: preventing a
deadlock. It's probably not correct as hinted by the FIXME comment but
better than deadlocking at least.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1701>
2024-08-13 06:04:09 +00:00
Mathieu Duponchelle
0da1c8e9c9 webrtcsink: fix assertions when finalizing
Dumping the pipeline on state changes from an async bus handler was
triggering criticals.

Instead, dump from the sync handler.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1706>
2024-08-12 09:13:06 +02:00
Sebastian Dröge
30a5987c9e rtp: mp4gpay: Don't set seqnum-base on the caps
This is supposed to be set by another layer, e.g. rtspsrc.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
Sebastian Dröge
de42ae432c rtp: basepay: Fix off-by-one with seqnum-offset
Setting a seqnum-offset of 1 would've caused the first packet to have a
seqnum of 2 instead of 1.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
Sebastian Dröge
c5163a73ee rtp: basepay: Don't negotiate twice in the beginning
If srcpad caps are already set as part of sinkpad caps handling, unset
the reconfigure flag so negotiation does not happen yet another time on
the first buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
Sebastian Dröge
31e836f4d6 rtp: basepay: Negotiate SSRC and PT with downstream if not set via property
This makes the new payloaders closer to the old ones, and makes usage in
webrtcbin easier.

Also properly configure default PT of subclasses. Previously any PT that
was set for these subclasses via g_object_new() would be overridden by
the default one during construction.

Additionally, do SSRC collision handling while queueing output packets.
This is the more natural place as that's where the SSRC is actually
used, it happens potentially earlier and also allows to drain any
pending packets before the SSRC change in the caps.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/557

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
Sebastian Dröge
914ffc8be9 rtp: basepay: Initialize class fields
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
Sebastian Dröge
c554a5dc76 rtp: basepay: Don't unset stats on FlushStop
They are still valid and unsetting them here would cause no stats to
ever be updated again until the next state change.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
Sebastian Dröge
035a199109 rtp: basepay: Don't use suggested SSRC on collissions if it's the current one
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1693>
2024-08-10 08:06:40 +00:00
Mathieu Duponchelle
9080c90120 net/webrtc: add support for answering to webrtcsink
Support was added to the base class when the AWS KVS signaller was
implemented, but the default signaller still only supported the case
where the producer was creating the offer.

Also extend the javascript API

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1702>
2024-08-09 14:02:48 +02:00
Mathieu Duponchelle
a9ff9615ff net/webrtc: correct signaller debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1702>
2024-08-08 18:28:43 +02:00
Mathieu Duponchelle
64f0b76f71 webrtc: update README with section on embedded signalling / web services
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-08-08 16:40:46 +02:00
Mathieu Duponchelle
9455e09d9f webrtcsink: expose properties for running web server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-08-08 16:40:46 +02:00
Mathieu Duponchelle
b709c56478 webrtcsink: expose properties for running signalling server
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1671>
2024-08-07 19:55:00 +02:00
Sebastian Dröge
6c04b59454 webrtcsrc: Don't hold the state lock while removing sessions
Removing a session can drop its bin and during release of the bin its
pads are removed, but the pad-removed handler is also taking the state
lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1695>
2024-08-07 09:35:15 +00:00
Sebastian Dröge
b83b6031e5 Update etherparse and async-tungstenite dependencies
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1691>
2024-08-06 09:00:32 +03:00
Dave Lucia
3a949db720 net/webrtc: Fix turn-servers nick: user -> use
Noticed this typo

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1690>
2024-08-05 12:38:51 -04:00
Sebastian Dröge
fa060b9fa0 Fix various 1.80 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1688>
2024-08-05 14:14:17 +03:00
Mathieu Duponchelle
86039dd5c1 webrtc-api example: do not rely on webpack / npm proxying websocket
Instead simply use the desired address directly from the reference
example, this makes it work out of the box without placing expectations
on the web server.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1674>
2024-07-30 16:29:54 +00:00
Loïc Le Page
5a1d12419f gstwebrtc-api: always include index file in dist for convenience
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1670>
2024-07-17 08:27:31 +00:00
François Laignel
34b791ff5e webrtc: add raw payload support
This commit adds support for raw payloads such as L24 audio to `webrtcsink` &
`webrtcsrc`.

Most changes take place within the `Codec` helper structure:

* A `Codec` can now advertise a depayloader. This also ensures that a format
  not only can be decoded when necessary, but it can also be depayloaded in the
  first place.
* It is possible to declare raw `Codec`s, meaning that their caps are compatible
  with a payloader and a depayloader without the need for an encoder and decoder.
* Previous accessor `has_decoder` was renamed as `can_be_received` to account
  for codecs which can be handled by an available depayloader with or without
  the need for a decoder.
* New codecs were added for the following formats:
  * L24, L16, L8 audio.
  * RAW video.

The `webrtc-precise-sync` examples were updated to demonstrate streaming of raw
audio or video.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1501>
2024-07-16 19:32:02 +00:00
Sebastian Dröge
d4d02d70a8 rtp: Require bitstream-io < 2.4.0
Version 2.4.0 contains a breaking change that it shouldn't, and updating
to 2.4.0 requires a newer Rust version.

See https://github.com/tuffy/bitstream-io/issues/22
2024-07-16 19:13:49 +03:00
François Laignel
6e9855c36b webrtcsink: fix property types for rav1enc
Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/572
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1667>
2024-07-12 18:59:20 +02:00
Sanchayan Maity
12be9a24a6 net/quinn: Fix generation of self signed certificate
The certificate chain was incorrectly being passed the private key instead
of certificate. With rustls 0.23.11 version, this error was being caught
and reported. As stated in the 0.23.11 release, it has a new feature

"API for determining whether a CertifiedKey's certificate and private key
matches: keys_match(). This is called from existing fallible functions
that accept a private key and certificate (for example, with_single_cert())
so these functions now detect this misconfiguration."

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1666>
2024-07-12 12:26:54 +05:30
Sebastian Dröge
a8ccfe49d9 webrtc: Require livekit-protocol < 0.3.4 due to uncoordinated breaking changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1663>
2024-07-11 20:00:24 +03:00
Sebastian Dröge
98b28d69ce Update for new debug log macro syntax
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1658>
2024-07-08 11:25:23 +03:00
Sebastian Dröge
4123b5d1a1 mpegtslive: Update for gst::Clock::set_calibration() API changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1658>
2024-07-08 09:59:06 +03:00
Sanchayan Maity
2fe852166e aws/s3hlssink: Do not call abort before finishing uploads
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1653>
2024-07-06 14:44:08 +00:00
Sebastian Dröge
4ab8d92f28 mpegtslivesrc: Don't skip the first MPEG-TS packet
If every buffer contains only a single MPEG-TS packet we would otherwise
skip over everything and would never observe a PCR.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1651>
2024-07-04 17:01:43 +03:00
Philippe Normand
eee93aea52 rtp2: Fix typo on auto-header-extension property name
The rtp (de)pay elements use auto-header-extension so the new elements should do
the same.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1646>
2024-07-02 09:35:39 +01:00
Edward Hervey
95ae67752f net: New mpegtslive element
This element allows wrapping an existing live "mpeg-ts source" (udpsrc,
srtsrc,...) and providing a clock based on the actual PCR of the stream.

Combined with `tsdemux ignore-pcr=True` downstream of it, this allows playing
back the content at the same rate as the (remote) provider **and** not modify
the original timestamps.

Co-authored-by: Sebastian Dröge <slomo@coaxion.net>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1640>
2024-07-01 15:29:22 +02:00
leonardo salvatore
f303992e0c webrtcsink: initial support for vpuenc_h264 encoder for imx8mp, default values set to cover a common streaming scenario
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1639>
2024-07-01 07:34:04 +00:00
Guillaume Desmottes
a10577b42c aws: log error if sink failed to start
I find it confusing that the element was failing without reporting any
error in its logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1638>
2024-06-26 11:22:54 +02:00
Guillaume Desmottes
0ecbd3f953 aws: use DisplayErrorContext when displaying SDK errors
As suggested in the aws crate documentation, wrap SDK errors with
DisplayErrorContext so their Display implementation outputs the full
context.

Improve error display from "dispatch failure" to

"dispatch failure: io error: error trying to connect: dns error: failed
to lookup address information: Name or service not known: dns error:
failed to lookup address information: Name or service not known: failed
to lookup address information: Name or service not known
(DispatchFailure(DispatchFailure { source: ConnectorError { kind: Io,
source: hyper::Error(Connect, ConnectError(\"dns error\", Custom { kind:
Uncategorized, error: \"failed to lookup address information: Name or
service not known\" })), connection: Unknown } }))"

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1638>
2024-06-26 10:47:10 +02:00
Guillaume Desmottes
3b7b2cd37b aws: rely on WaitError Display implementation
The Display implementation of WaitError already displays the underlying
SDK error and the metadata, so can just use that.

Will also be used to provide more context in the next patch.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1638>
2024-06-26 10:46:46 +02:00
Sanchayan Maity
0bd98e2c34 net/quinn: Allow dropping buffers when buffer size exceeds maximum datagram size
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
e00ebca63f net/quinn: Add stats property for connection statistics
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
2b35f009fb net/quinn: Update quinn to 0.11.2
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
cf7172248c net/quinn: Allow setting some parameters from TransportConfig
As of now, we expose the below four properties from `TransportConfig`.
- Initial MTU
- Minimum MTU
- Datagram receive buffer size
- Datagram send buffer size

Maximum UDP payload size from `EndpointConfig` and upper bound from
`MtuDiscoveryConfig` are also exposed as properties.

See the below documentation for further details.
- https://docs.rs/quinn/latest/quinn/struct.TransportConfig.html
- https://docs.rs/quinn/latest/quinn/struct.MtuDiscoveryConfig.html
- https://docs.rs/quinn/latest/quinn/struct.EndpointConfig.html

While at it, also clean up passing function parameters to the functions
in utils.rs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Sanchayan Maity
bc5ed023e4 net/quinn: Improve datagram handling
We now check if the peer actually supports Datagram and refusing to
proceed if it does not. Since the datagram size can actually change
over the lifetime of a connection according to variation in path MTU
estimate, also check buffer size before trying to send.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1613>
2024-06-25 20:15:40 +05:30
Matthew Waters
39b61195ad rtprecv: ensure that stopping the rtp src task does not critical
When pad a released, then we were removing the pad from an internal
list. If the pad was not already deactivated, the deactiviation would
attempt to look for the pad in that list and panic if it was not there.

Fix by delaying removal of the pad from the list until after pad
deactivation occurs.

Also includes test.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
10a31a397e rtp/recv: support pushing buffer lists from the jitterbuffer
Multiple concurrent buffers produced by the jitterbuffer will be
combined into a single buffer list which will be sent downstream.

Events or queries that interrupt the buffer flow will cause a split in
the output buffer list.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
d036abb7d2 rtp/recv: support buffers lists on rtp sink pad
In one case, improves throughput by 25% when buffer lists are used.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
df4a4fb2ef rtp/send: support receiving buffer lists
Can reduce processing overhead if many buffers are pushed concurrently.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
2d1f556794 rtp/session: guard against a busy wait with no members
If the number of members is 0, then the calculated time to the next rtcp
wakup would be 'now' and could result in a busy loop in the rtcp
processing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Matthew Waters
84a9f9c61f rtp/source: use extended sequence number helper
Instead of rolling our own

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1618>
2024-06-24 13:13:28 +00:00
Sebastian Dröge
47d62b6d78 Update for new clone/closure macro syntax
Also fix various weak/strong references in the webrtc plugin, and make
sure to pass the object to debug log functions in every place.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1625>
2024-06-21 11:54:58 +03:00
Sebastian Dröge
9b323a6519 Use Option::is_some_and(...) instead of Option::map_or(false, ...)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
2024-06-19 13:03:37 +00:00
Sebastian Dröge
23d998a1db Slightly improve code making use of element factories retrieved from an element
We can use `is_some_and(...)` instead of `map_or(false, ...)`.

Also in a few places the factory was retrieved multiple times, one time
with unwrapping and another time with handling the `None` case
correctly. Instead of unwrapping, move code to handle the `None` case.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1630>
2024-06-19 13:03:37 +00:00
Arun Raghavan
8f96509f03 aws: s3: Enable tests again
We lost the environment variable checks during the addition of the
putobjectsink tests, which caused failures on MR branches.

It would be nicer to use some other mechanism to validate the tests can
run, so we don't count on only the environmnent, but for now this will
have to do.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1629>
2024-06-18 11:58:43 -04:00
Sebastian Dröge
743ab29ba8 Update Cargo.lock and MSRV to 1.71
cea608-types requires that now because it updated the env_logger
dependency. As a result, we can also update it here now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1622>
2024-06-18 10:27:27 +03:00
Sebastian Dröge
5aedcab32f Revert "aws: s3: Re-enable tests"
This reverts commit b4b56eb282.
The tests are still failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:50:07 +03:00
Sebastian Dröge
4677948a82 rtp: av1pay: Derive Default trait for the state instead of manual implementation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:07:24 +03:00
Sebastian Dröge
d357a63bf9 rtp: av1pay: Correctly use N flag for marking keyframes
The "first packet of a coded video sequence" means that this should be
the first packet of a keyframe that comes together with a sequence
header, not the first packet of a new frame.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/558

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:06:59 +03:00
Sebastian Dröge
5cd9e34265 rtp: av1pay: Correctly skip over ignored OBUs
The reader is already after the header at this point so only the OBU
content has to be skipped.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:06:59 +03:00
Sebastian Dröge
bbe38b9599 rtp: av1: Drop padding OBUs too like Chrome does
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1624>
2024-06-18 08:06:59 +03:00
Arun Raghavan
b4b56eb282 aws: s3: Re-enable tests
These seem to have stopped working due to bad/rotated creds. Should work
fine now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1623>
2024-06-17 06:08:18 -04:00
Sebastian Dröge
343680ffea rtp: av1depay: Don't return an error if parsing a packet fails
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
2024-06-14 13:13:21 +00:00
Sebastian Dröge
477855789d rtp: av1depay: Also log warnings on errors
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
2024-06-14 13:13:21 +00:00
Sebastian Dröge
93c9821cba rtp: av1depay: Drop unusable packets as early as possible
Otherwise they would pile up until a discontinuity or until we can
actually output something.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
2024-06-14 13:13:21 +00:00
Sebastian Dröge
0ca4a3778a rtp: av1depay: Parse internal size fields of OBUs and handle them
They're not recommended by the spec to include in the RTP packets but it
is valid to include them. Pion is including them.

When parsing the size fields also make sure to only take that much of a
payload unit and to skip any trailing data (which should not exist in
the first place).

Pion is also currently storing multiple OBUs in a single payload unit,
which is not allowed by the spec but can be easily handled with this
code now.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/560

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1612>
2024-06-14 13:13:21 +00:00
Sebastian Dröge
69c3c2ae46 Fix various new clippy 1.79 warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1620>
2024-06-14 08:33:49 +03:00
Sebastian Dröge
3d4d785a2a webrtchttp: Fix race condition when unlocking
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if cancellation should happen and reset this
later.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
2024-06-10 07:38:29 +00:00
Sebastian Dröge
51f6d3986f aws: Fix race condition when unlocking
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if unlock() was called and reset this in
unlock_stop().

Also implement actual unlocking in s3hlssink.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
2024-06-10 07:38:29 +00:00
Sebastian Dröge
00aaecad07 quinn: Fix race condition when unlocking
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if unlock() was called and reset this in
unlock_stop().

Also actually implement unlock() / unlock_stop() for the sink, and don't
cancel in stop() as unlock() / unlock_stop() would've been called before
that already.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
2024-06-10 07:38:29 +00:00
Sebastian Dröge
9945b702b8 reqwesthttpsrc: Fix race condition when unlocking
It would be possible that there is no cancellable yet when unlock() is
called, then a new future is executed and it wouldn't have any
information that it is not supposed to run at all.

To solve this remember if unlock() was called and reset this in
unlock_stop().

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1602>
2024-06-10 07:38:29 +00:00
Sebastian Dröge
f68655b5e2 Update for gst::BufferList API changes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1610>
2024-06-08 09:58:10 +03:00
Sebastian Dröge
30252a1b2e ndi: Add support for loading NDI SDK v6
The library name and environment variable name have changed but the ABI
is completely compatible.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1607>
2024-06-06 14:51:09 +00:00
Matthew Waters
260b04a1cf rtpbin2: protoct against adding with overflow
If jitter is really bad, then this calculation may overflow.  Protect
against that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1605>
2024-06-06 11:43:26 +00:00
Sebastian Dröge
ba70bb1154 deny: Add override for older tungstenite
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
2024-06-06 10:34:12 +00:00
Sebastian Dröge
85c38107cf webrtc: Update to async-tungstenite 0.26
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1603>
2024-06-06 10:34:12 +00:00
Sanchayan Maity
8171a00943 net/quinn: Fix pad template naming typo
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1601>
2024-06-05 13:44:40 +05:30
Tim-Philipp Müller
ab2f5e3d8d rtp: ac3: add some unit tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
2b68920f82 rtp: tests: add possibility to make input live
.. for payloaders that behave differently with live
and non-live inputs (e.g. audio payloaders which by
default will pick different aggregation modes based
on that.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
6597ec84eb rtp: tests: add possibility to check duration of depayloaded buffers
.. and clarify an expect panic message.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
2024-06-01 12:43:27 +00:00
Tim-Philipp Müller
6b628485c5 rtp: Add AC-3 RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1586>
2024-06-01 12:43:27 +00:00
Tamas Levai
802ff6a67c net/quinn: Make QUIC role configurable
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1575>
2024-05-31 23:20:38 +02:00
Francisco Javier Velázquez-García
8fc652f208 webrtcsink: Refactor value retrieval to avoid lock poisoning
When setting an incorrect property name in settings,
start_stream_discovery_if_needed would panic because it attempts to
unwrap a poisoned lock for settings.

This refactor avoids that situation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
2024-05-31 08:10:23 +00:00
Francisco Javier Velázquez-García
568e8533fa webrtcsink: Fix typo in property name for av1enc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1594>
2024-05-31 08:10:23 +00:00
Arun Raghavan
04e9e5284c webrtc: signaller: A couple of minor doc fixups
The expectation is `Returns:`, not `Return:`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
2024-05-30 22:16:46 +03:00
Arun Raghavan
1c54c77840 webrtcsink: Add a mechanism for SDP munging
Unfortunately, server implementations might have odd SDP-related quirks,
so let's allow clients a way to work around these oddities themselves.
For now, this means that a client can fix up the H.264 profile-level-id
as required by Twitch (whose media pipeline is more permissive than the
WHIP implementation).

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/516
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1525>
2024-05-30 22:16:46 +03:00
Taruntej Kanakamalla
83f76280f5 net/webrtc: Example for whipserver
rudimentary sample to test multiple WHIP client connections

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla
712d4757c3 net/webrtc/whip_signaller: multiple client support in the server
- generate a new session id for every new client
use the session id in the resource url

- remove the producer-peer-id property in the WhipServer signaler as it
is redundant to have producer id in a session having only one producer

- read the 'producer-peer-id' property on the signaller conditionally
if it exists else use the session id as producer id

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2024-05-29 21:03:27 +00:00
Taruntej Kanakamalla
de726ca8d2 net/webrtc: multi producer support in webrtcsrc
- Add a new structure Session
  - manage each producer using a session
  - avoid send EOS when a session terminates, instead keep running
    waiting for any new producer to connect

- Maintain a bin element per session
  - each session bin encapsulates webrtcbin and the decoder if needed
    as well as the parser and filter if requested by the application
    (through request-encoded-filter)
  - this will be helpful to cleanup the session's respective elements
    when the corresponding producer terminates the session

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1339>
2024-05-29 21:03:27 +00:00
Sebastian Dröge
a7418fb483 rtp: Use released version of rtcp-types 2024-05-29 10:30:40 +03:00
Matthew Waters
df32e1ebfa rtpsend: ensure only a single rtcp pad push
Otherwise, it can occur that multiple rtcp packets may be produced out
of order.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Matthew Waters
525179f666 rtpbin2: handle ssrc collisions
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Nirbheek Chauhan
9485265769 rtspsrc2: Update rtpbin2 support to use rtprecv and rtpsend
USE_RTPBIN2 is now USE_RTP2 because there is no "rtpbin2" now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Matthew Waters
1600d3b055 rtpbin2: split send and receive halves into separate elements
There is now two elements, rtpsend and rtprecv that represent the two
halves of a rtpsession.  This avoids the potential pipeline loop if two
peers are sending/receiving data towards each other.  The two halves can
be connected by setting the rtp-id property on each element to the same
value and they will behave like a combined rtpbin-like element.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Matthew Waters
0121d78482 rtpbin2: expose session signals for new/bye ssrc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Matthew Waters
d480c6c2d3 rtpbin2/config: add stats to session GObject
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Matthew Waters
7d5789032a rtpbin2/config: add a new-ssrc signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Matthew Waters
06f40e72cb rtpbin2: implement a session configuration object
Currently only contains pt-map

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00