Commit graph

897 commits

Author SHA1 Message Date
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
Matthew Waters
48e7a2ed06 jitterbuffer: handle flush-start/stop
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Matthew Waters
66306e32f2 jitterbuffer: remove mpsc channel for every packet
It is very slow.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
327f563e80 jitterbuffer: implement support for serialized events / queries
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
74ec83a0ff rtpbin2: implement and use synchronization context
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 19:58:09 +10:00
Mathieu Duponchelle
1865899621 rtpbin2: implement jitterbuffer
The jitterbuffer implements both reordering and duplicate packet
handling.

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 17:35:41 +10:00
Sebastian Dröge
2b4ec75bc5 rtpbin2: Add support for receiving rtcp-mux packets
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 17:35:41 +10:00
Sebastian Dröge
e09ad990fa rtpbin2: Implement support for reduced size RTCP (RFC 5506)
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 17:35:41 +10:00
Sebastian Dröge
1e4a966c92 rtpbin2: Add support for sending NACK/PLI and FIR
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 17:35:41 +10:00
Sebastian Dröge
66c9840ad8 rtpbin2: Add handling for receiving NACK/PLI and FIR
Co-Authored-By: Matthew Waters <matthew@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 17:35:41 +10:00
Matthew Waters
2c86f18a99 rtpbin2: add support for RFC 4585 (RTP/AVPF)
Implements the timing rules for RTP/AVPF

Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 17:35:41 +10:00
Matthew Waters
27ad26c258 rtp: Initial rtpbin2 element
Can receive and recevie one or more RTP sessions containing multiple
pt/ssrc combinations.

Demultiplexing happens internally instead of relying on separate
elements.

Co-Authored-By: François Laignel <francois@centricular.com>
Co-Authored-By: Mathieu Duponchelle <mathieu@centricular.com>
Co-Authored-By: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1426>
2024-05-28 17:35:41 +10:00
Sebastian Dröge
984a9fe5ff rtp: Don't restrict payload types for payloaders
WebRTC uses payload types 35-63 as dynamic payload types too to be able
to place more codec variants into the SDP offer.

Instead of allowing just certain payload types, completely remove any
restrictions and let the user decide. There's technically nothing wrong
with using any payload type, especially when using the encoding-name.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/551

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1587>
2024-05-27 09:34:16 +00:00
Liam
b4fd6cf362 aws: Add system-defined metadata options to both sinks
Add to awss3sink and awss3putobjectsink elements the following
paramerters which are set on the uploaded S3 objects:

* cache-control;
* content-encoding; and
* content-language

Bugfix: Set the content-type and content-disposition values in the S3
putobject call. Previously the params were defined on the element but
unused.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1585>
2024-05-27 10:25:22 +03:00
Tim-Philipp Müller
4f74cb7958 rtp: klv: add test for fragmented payloads with packet loss
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
b6e24668a7 rtp: klv: add unit test with some packet loss
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
92a1e222f4 rtp: tests: add functionality to drop RTP packets after payloading
Add ExpectedPacket::drop() to flag RTP packets that should not
be forwarded to the depayloader.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
de71e9dadd rtp: tests: print rtp timestamp mismatch minus the initial offset
Unit tests specify a 0-based offset, so printing that plus the
random initial offset on failure is just needlessly confusing,
so subtract the initial offset when printing expected/actual
values. The real values are still printed as part of the assert.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
be7da027f8 rtp: klv: add some basic tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
1e33926dc5 fixup: klv payloader indentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
c2f67bd3c9 fixup: klv depay: debug log indentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
e7d0e0702a fixup: payloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-26 12:34:44 +03:00
Tim-Philipp Müller
566e6443f4 rtp: Add KLV RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1580>
2024-05-25 20:21:50 +03:00
François Laignel
4259d284bd webrtc: add android webrtcsrc example
This commit adds an Android `webrtcsrc` based example with the following
features:

* A first view allows retrieving the producer list from the signaller (peer ids
  are uuids which are too long to tap, especially using an onscreen keyboard).
* Selecting a producer opens a second view. The first available video stream is
  rendered on a native Surface. All the audio streams are rendered using
  `autoaudiosink`.

Available Settings:

* Signaller URI.
* A toggle to prefer hardware decoding for OPUS, otherwise the app defaults to
  raising `opusdec`'s rank. Hardware decoding was moved aside since it was found
  to crash the app on all tested devices (2 smartphones, 1 tv).

**Warning**: in order to ease testing, this demonstration application enables
unencrypted network communication. See `AndroidManifest.xml`.

The application uses the technologies currenlty proposed by Android Studio when
creating a new project:

* Kotlin as the default language, which is fully interoperable with Java and
  uses the same SDK.
* gradle 8.6.
* kotlin dialect for gradle. The structure is mostly the same as the previously
  preferred dialect, for which examples can be found online readily.
* However, JNI code generation still uses Makefiles (instead of CMake) due to
  the need to call [`gstreamer-1.0.mk`] for `gstreamer_android` generation.
  Note: on-going work on that front:
  - https://gitlab.freedesktop.org/gstreamer/cerbero/-/merge_requests/1466
  - https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6794

Current limitations:

* x86 support is currently discarded as `gstreamer_android` libs generation
  fails (observed with `gstreamer-1.0-android-universal-1.24.3`).
* A selector could be added to let the user chose the video streams and
  possibly decide whether to render all audio streams or just select one.

Nice to have:

* Support for the synchronization features of the `webrtc-precise-sync-recv`
  example (NTP clock, RFC 7273).
* It could be nice to use Rust for the specific native code.

[`gstreamer-1.0.mk`]: https://gitlab.freedesktop.org/gstreamer/cerbero/-/blob/main/data/ndk-build/gstreamer-1.0.mk

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1578>
2024-05-24 16:14:13 +00:00
Sebastian Dröge
58e91c154c rtp: basedepay: Reset last used ext seqnum on discontinuities
The ext seqnum counting is reset too so keeping the old one around will
cause problems with timestamping of the next outgoing buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1584>
2024-05-24 10:23:06 +03:00
cdelguercio
c99cabfbc5 webrtcsink: Add VP9 parser after the encoder for VP9 too
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-23 10:16:59 +03:00
cdelguercio
f5a7de9dc3 webrtcsink: Support av1 via nvav1enc, av1enc, and rav1enc
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1572>
2024-05-23 10:16:59 +03:00
Sebastian Dröge
dcc0b47349 rtp: basepay: Fix header extension negotiation
Only configure header extensions from the source pad caps if they exist
already in the source pad caps, otherwise the configuration will fail.
Extensions that are added via the signals might not exist in the source
pad caps yet and would be added later.

Also, if configuring an existing extension from the new caps fails,
remove it and try to request a new extension for it.

Additionally don't remove extensions from the caps that can't be
provided. No header extensions for them would be added to the packets,
but that's not a problem. Removing them on the other hand would cause
negotiation to fail. This only affects extensions that are already
included in the caps.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
2024-05-20 07:53:50 +00:00
Sebastian Dröge
0d33077df6 rtp: basedepay: Clean up header extension negotiation
If configuring an existing extension from the new caps fails, remove it
and try to request a new extension for it.

Also remove all extensions from the list that are not provided in the
caps, instead of passing RTP packets to all of them anyway.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1577>
2024-05-20 07:53:50 +00:00
Tim-Philipp Müller
16608d2541 rtp: opus: add multichannel depay/pay test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
bab3498c6a rtp: opus: add multichannel pay/depay test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
72006215cb rtp: tests: add run_test_pipeline_full() that checks output caps too
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
10e0294d5a rtp: opus: fix payloader caps query handling and add tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
61523baa7b rtp: opus: add minimal depayload / re-payload test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
6f871e6ce2 rtp: opus: add simple payload / depayload test
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
92c0cf1285 rtp: opus: add test for payloader dtx packet handling
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Tim-Philipp Müller
2585639054 rtp: Add Opus RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1571>
2024-05-18 09:29:29 +00:00
Sebastian Dröge
539000574b aws: Update to base32 0.5
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1576>
2024-05-17 07:50:51 +00:00
Robert Ayrapetyan
bac5845be1 webrtc: add support for insecure tls connections
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1553>
2024-05-16 19:34:57 +00:00
Martin Nordholts
9a7f37e2b7 rtpgccbwe: Support linear regression based delay estimation
In our tests, the slope (found with linear regression) on a
history of the (smoothed) accumulated inter-group delays
gives a more stable congestion control. In particular,
low-end devices becomes less sensitive to spikes in
inter-group delay measurements.

This flavour of delay based bandwidth estimation with Google
Congestion Control is also what Chromium is using.

To make it easy to experiment with the new estimator, as
well as add support for new ones in the future, also add
infrastructure for making delay estimator flavour selectable
at runtime.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
2024-05-14 16:25:48 +00:00
Martin Nordholts
71e9c2bb04 rtpgccbwe: Also log self.measure in overuse_filter()
Also log `self.measure` in overuse_filter() since tracking
`self.measure` over time help a lot in making sense of
`self.estimate` (and `amplified_estimate`).

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
2024-05-14 16:25:48 +00:00
Martin Nordholts
d9aa0731f4 rtpgccbwe: Rename variable t to amplified_estimate
We normally multiply `self.estimate` with `MAX_DELTAS` (60).
Rename the variables that holds the result of this
calculation to `amplified_estimate` to make the distinction
clearer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1566>
2024-05-14 16:25:48 +00:00
Tamas Levai
71cd80f204 net/quinn: Enable client to keep QUIC conn alive
Co-authored-by: Felician Nemeth <nemethf@tmit.bme.hu>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1568>
2024-05-11 08:51:00 +02:00
Sebastian Dröge
613ed56675 webrtcsink: Add a custom signaller example in Python
This re-implements the default webrtcsink/src signalling protocol in
Python for demonstration purposes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1569>
2024-05-10 15:59:12 +00:00
Martin Nordholts
a719cbfcc6 rtp: Change RtpBasePay2::ssrc_collision from AtomicU64 to Option<u32>
Rust targets without support for `AtomicU64` is still
somewhat common. Running

    git grep -i 'max_atomic_width: Some(32)' | wc -l

in the Rust compiler repo currently counts to 34 targets.

Change the `RtpBasePay2::ssrc_collision` from `AtomicU64` to
`Mutex<Option<u32>>`. This way we keep support for these
targets.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1562>
2024-05-10 14:23:41 +00:00
Martin Nordholts
aabb011f5a rtpgccbwe: Log effective bitrate in more places
While monitoring and debugging rtpgccbwe, it is very helpful
to get continuous values of what it considers the effective
bitrate. Right now such prints will stop coming once the
algorithm stabilizes. Print it in more places so it keeps
coming. Use the same format to make it simpler to extract
the values by parsing the logs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1567>
2024-05-10 11:56:51 +00:00
Martin Nordholts
e845e3575c rtpgccbwe: Add mising 'ps' suffix to 'kbps' of 'effective bitrate'
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1567>
2024-05-10 11:56:51 +00:00
Sebastian Dröge
e8e173d0d0 webrtc: Update Signallable interface to new interface definition API
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1570>
2024-05-10 14:13:55 +03:00
Sebastian Dröge
7e09481adc rtp: Add JPEG RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1543>
2024-05-10 11:12:49 +03:00
Sanchayan Maity
fe55acb4c9 net/hlssink3: Refactor out HlsBaseSink & hlscmafsink from hlssink3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1564>
2024-05-09 21:50:32 +05:30
Tamas Levai
5884c00bd0 net/quinn: Improve stream shutdown process
Co-authored-by: Sanchayan Maity <sanchayan@asymptotic.io>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1565>
2024-05-09 16:43:26 +02:00
Tamas Levai
13c3db7857 net/quinn: Port to quinn 0.11 and rustls 0.23
Co-authored-by: Felician Nemeth <nemethf@tmit.bme.hu>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1565>
2024-05-09 13:49:33 +02:00
Martin Nordholts
2b7488a4c8 rtpgccbwe: Log delay and loss target bitrates separately
When debugging rtpgccbwe it is helpful to know if it is
delay based or loss based band-width estimation that puts a
bound on the current target bitrate, so add logs for that.

To minimize the time we need to hold the state lock, perform
the logging after we have released the state lock.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1561>
2024-05-08 19:12:44 +00:00
Mathieu Duponchelle
8861fc493b webrtcsink: improve error when no discovery pipeline runs
If for instance no encoder was found or the RTP plugin was missing,
it is possible that no discovery pipeline will run for a given stream.

Provide a more helpful error message for that case.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/534
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1560>
2024-05-06 11:39:48 +00:00
Sanchayan Maity
3a3cec96ff net/quinn: Add pipeline example
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1558>
2024-05-02 16:39:29 +00:00
Sanchayan Maity
80f8664564 net/quinn: Use camel case acronym
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1558>
2024-05-02 16:39:29 +00:00
Sebastian Dröge
be3ae583bc Fix new Rust 1.78 clippy warnings
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1559>
2024-05-02 18:36:23 +03:00
Sebastian Dröge
58106a42a9 quinn: Fix up dependencies 2024-05-02 09:59:55 +03:00
Sanchayan Maity
150ad7a545 net/quinn: Use separate property for certificate & private key file
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 22:30:23 +05:30
Sanchayan Maity
0d2f054c15 Move net/quic to net/quinn
While at it, add this to meson.build.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 22:30:23 +05:30
Sanchayan Maity
18cf5292b7 net/quic: Fix inconsistencies around secure connection handling
This set of changes implements the below fixes:

- Allow certificates to be specified for client/quicsink
- Secure connection being true on server/quicsrc and false on
  client/quicsink still resulted in a successful connection
  instead of server rejecting the connection
- Using secure connection with ALPN was not working

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:09:16 +05:30
Sanchayan Maity
97d8a79d36 net/quic: Drop private key type property
Use read_all helper from rustls_pemfile and drop the requirement for the
user having to specify the private key type.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:09:16 +05:30
Sanchayan Maity
a306b1ce94 net/quic: Use a custom ALPN string
`h3` does not make sense as the default ALPN, as there likely isn't
going to be a HTTP/3 application layer, especially as our transport
is unidirectional for now. Use a custom string `gst-quinn` for now.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:09:16 +05:30
Sanchayan Maity
22c6a98914 net/quic: Rename to quinnquicsink/src
There might be other QUIC elements in the future based on other
libraries. To prevent namespace collision, namespace the elements
with `quinn` prefix.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:09:16 +05:30
Sanchayan Maity
8b64c734e7 net/quic: Use separate property for address and port
While at it, do not duplicate call to settings lock in property
getter and setter for every property.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:01:49 +05:30
Tamas Levai
befd8d4bd2 net/quic: Allow SSL keylog file for debugging
rustls has a KeyLog implementation that opens a file whose name is
given by the `SSLKEYLOGFILE` environment variable, and writes keys
into it. If SSLKEYLOGFILE is not set, this does nothing.

See
https://docs.rs/rustls/latest/rustls/struct.KeyLogFile.html
https://docs.rs/rustls/latest/rustls/trait.KeyLog.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:01:49 +05:30
Sanchayan Maity
ce930eab5f net/quic: Allow setting multiple ALPN transport parameters
For reference, see
https://datatracker.ietf.org/doc/html/rfc9000#section-7.4
https://datatracker.ietf.org/doc/html/rfc7301

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:01:49 +05:30
Tamas Levai
75b25d011f net/quic: Allow specifying an ALPN transport parameter
See https://datatracker.ietf.org/doc/html/rfc9000#section-7.4.

This controls the Transport Layer Security (TLS) extension for
application-layer protocol negotiation within the TLS handshake.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:01:49 +05:30
Sanchayan Maity
953f6a3fd7 net: Add QUIC source and sink
To test, run receiver as

```bash
gst-launch-1.0 -v -e quicsrc caps=audio/x-opus use-datagram=true ! opusparse ! opusdec ! audio/x-raw,format=S16LE,rate=48000,channels=2,layout=interleaved ! audioconvert ! autoaudiosink
```

run sender as

```bash
gst-launch-1.0 -v -e audiotestsrc num-buffers=512 ! audio/x-raw,format=S16LE,rate=48000,channels=2,layout=interleaved ! opusenc ! quicsink use-datagram=true
```

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1036>
2024-05-01 18:01:49 +05:30
François Laignel
16b0a4d762 rtp: add mp4gpay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
b588ee59bc rtp: add mp4gdepay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
5466cafc24 rtp: add mp4apay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
François Laignel
812fe0a9bd rtp: add mp4adepay
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1551>
2024-04-29 13:33:42 +00:00
Sebastian Dröge
7573caa8e9 rtpgccbwe: Move away from deprecated time::Instant to std::time::Instant
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1554>
2024-04-25 15:37:28 +03:00
Sebastian Dröge
d6a855ff1b rtp: Add VP8/9 RTP payloader/depayloader
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1487>
2024-04-15 14:03:56 +00:00
François Laignel
542030fd82 webrtcsink: don't panic if input CAPS are not supported
If a user constrained the supported CAPS, for instance using `video-caps`:

```shell
gst-launch-1.0 videotestsrc ! video/x-raw,format=I420 ! x264enc \
    ! webrtcsink video-caps=video/x-vp8
```

... a panic would occur which was internally caught without the user being
informed except for the following message which was written to stderr:

> thread 'tokio-runtime-worker' panicked at net/webrtc/src/webrtcsink/imp.rs:3533:22:
>   expected audio or video raw caps: video/x-h264, [...] <br>
> note: run with `RUST_BACKTRACE=1` environment variable to display a backtrace

The pipeline kept running.

This commit converts the panic into an `Error` which bubbles up as an element
`StreamError::CodecNotFound` which can be handled by the application.
With the above `gst-launch`, this terminates the pipeline with:

> [...] ERROR  webrtcsink net/webrtc/src/webrtcsink/imp.rs:3771:gstrswebrtc::
>   webrtcsink:👿:BaseWebRTCSink::start_stream_discovery_if_needed::{{closure}}:<webrtcsink0>
> Error running discovery: Unsupported caps: video/x-h264, [...] <br>
> ERROR: from element /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
>   There is no codec present that can handle the stream's type. <br>
> Additional debug info: <br>
> net/webrtc/src/webrtcsink/imp.rs(3772): gstrswebrtc::webrtcsink:👿:BaseWebRTCSink::
> start_stream_discovery_if_needed::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSink:webrtcsink0:
> Failed to look up output caps: Unsupported caps: video/x-h264, [...] <br>
> Execution ended after 0:00:00.055716661 <br>
> Setting pipeline to NULL ...

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1540>
2024-04-14 23:09:09 +02:00
François Laignel
3fc38be5c4 webrtc: add missing tokio feature for precise sync examples
Clippy caught the missing feature `signal` which is used by the WebRTC precise
synchronization examples. When running `cargo` `check`, `build` or `clippy`
without `no-default-dependencies`, this feature was already present due to
dependents crates.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1541>
2024-04-14 16:50:33 +02:00
François Laignel
168af88eda webrtc: add features for specific signallers
When swapping between several development branches, compilation times can be
frustrating. This commit proposes adding features to control which signaller
to include when building the webrtc plugin. By default, all signallers are
included, just like before.

Compiling the `webrtc-precise-sync` examples with `--no-default-features`
reduces compilation to 267 crates instead of 429 when all signallers are
compiled in.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1539>
2024-04-12 19:10:42 +02:00
François Laignel
83d70d3471 webrtc: add RFC 7273 support
This commit implements [RFC 7273] (NTP & PTP clock signalling & synchronization)
for `webrtcsink` by adding the "ts-refclk" & "mediaclk" SDP media attributes to
identify the clock. These attributes are handled by `rtpjitterbuffer` on the
consumer side. They MUST be part of the SDP offer.

When used with an NTP or PTP clock, "mediaclk" indicates the RTP offset at the
clock's origin. Because the payloaders are not instantiated when the offer is
sent to the consumer, the RTP offset is set to 0 and the payloader
`timstamp-offset`s are set accordingly when they are created.

The `webrtc-precise-sync` examples were updated to be able to start with an NTP
(default), a PTP or the system clock (on the receiver only). The rtp jitter
buffer will synchronize with the clock signalled in the SDP offer provided the
sender is started with `--do-clock-signalling` & the receiver with
`--expect-clock-signalling`.

[RFC 7273]: https://datatracker.ietf.org/doc/html/rfc7273

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1500>
2024-04-12 14:18:09 +02:00
François Laignel
42158cbcb0 gccbwe: don't log an error when handling a buffer list while stopping
When `webrtcsink` was stopped, `gccbwe` could log an error if it was handling a
buffer list. This commit logs an error only if `push_list()` returned an error
other than `Flushing`.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1535>
2024-04-11 01:29:53 +00:00
Sanchayan Maity
a3e30b499f aws: Introduce a property to use path-style addressing
AWS SDK switched to virtual addressing as default instead of path
style earlier. While MinIO supports virtual host style requests,
path style requests are the default.

Introduce a property to allow the use of path style addressing if
required.

For more information, see
https://github.com/minio/minio/blob/master/docs/config/README.md#domain
https://docs.aws.amazon.com/AmazonS3/latest/userguide/VirtualHosting.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1527>
2024-04-10 00:23:22 +00:00
François Laignel
2ad452ee89 webrtcsink: don't panic with bitrate handling unsupported encoders
When an encoder was not supported by the `VideoEncoder` `bitrate` accessors, an
`unimplemented` panic would occur which would poison `state` & `settings`
`Mutex`s resulting in other threads panicking, notably entering `end_session()`,
which lead to many failures in `BinImplExt::parent_remove_element()` until a
segmentation fault ended the process. This was observed using `vaapivp9enc`.

This commit logs a warning if an encoder isn't supported by the `bitrate`
accessors and silently by-passes `bitrate`-related operations when unsupported.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1534>
2024-04-09 15:48:59 +00:00
Taruntej Kanakamalla
f4b086738b webrtcsrc: change the producer-id type for request-encoded-filter
With https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1477
the producer id used while emitting the request-encoded-filter
can be a None if the msid of the webrtcbin's pad is None.
This might not affect the signal handler written in C but
can panic in an existing Rust application with signal
handler which can only handle valid String type as its param
for the producer id.

So change the param type to Option<String> in the signal builder
for request-encoded-fiter signal

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1528>
2024-04-09 06:01:15 +00:00
Sebastian Dröge
fab246f82e webrtchttp: Update to reqwest 0.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1530>
2024-04-06 11:07:16 +03:00
Sebastian Dröge
7757e06e36 onvifmetadataparse: Reset state in PAUSED->READY after pad deactivation
Otherwise the clock id will simply be overridden instead of unscheduling
it, and if the streaming thread of the source pad currently waits on it
then it will wait potentially for a very long time and deactivating the
pad would wait for that to happen.

Also unschedule the clock id on `Drop` of the state to be one the safe
side and not simply forget about it.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1526>
2024-04-05 15:19:37 +00:00
Taruntej Kanakamalla
70adedb142 net/webrtc: fix inconsistencies in documentation of object names
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1529>
2024-04-05 14:10:35 +00:00
François Laignel
cc43935036 webrtc: add precise synchronization example
This example demonstrates a sender / receiver setup which ensures precise
synchronization of multiple streams in a single session.

[RFC 6051]-style rapid synchronization of RTP streams is available as an option.
See the [Instantaneous RTP synchronization...] blog post for details about this
mode and an example based on RTSP instead of WebRTC.

[RFC 6051]: https://datatracker.ietf.org/doc/html/rfc6051
[Instantaneous RTP synchronization...]: https://coaxion.net/blog/2022/05/instantaneous-rtp-synchronization-retrieval-of-absolute-sender-clock-times-with-gstreamer/

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1463>
2024-04-03 19:10:40 +02:00
Guillaume Desmottes
b5cbc47cf7 web: webrtcsink: improve panic message on unexpected caps during discovery
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
2024-04-02 14:25:58 +02:00
Guillaume Desmottes
35b84d219f webrtc: webrtcsink: set perfect-timestamp=true on audio encoders
Chrome audio decoder doesn't cope well with not perfect ts, generating
noises in the audio.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1524>
2024-04-02 14:25:51 +02:00
Martin Nordholts
5d7e068a8b rtpgccbwe: Add increasing_duration and counter to existing gst::log!()
Add `self.increasing_duration` and `self.increasing_counter`
to logs to provide more details of why `overuse_filter()`
determines overuse of network.

To get access to the latest values of those fields we need
to move down the log call. But that is fine, since no other
logged data is modified between the old and new location of
`gst::log!()`.

We do not bother logging `self.last_overuse_estimate` since
that is simply the previously logged value of `estimate`. We
must put the log call before we write the latest value to it
though, in case we want to log it in the future.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1522>
2024-03-27 15:08:23 +00:00
François Laignel
a870d60621 aws: improve error message logs
The `Display` and `Debug` trait for the AWS error messages are not very useful.

- `Display` only prints the high level error, e.g.: "service error".
- `Debug` prints all the fields in the error stack, resulting in hard to read
  messages with redudant or unnecessary information. E.g.:

> ServiceError(ServiceError { source: BadRequestException(BadRequestException {
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), meta: ErrorMetadata { code: Some("BadRequestException"),
> message: Some("1 validation error detected: Value 'test' at 'languageCode'
> failed to satisfy constraint: Member must satisfy enum value set: [ar-AE,
> zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH,
> de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US,
> fr-CA, en-GB]"), extras: Some({"aws_request_id": "1b8bbafd-5b71-4ba5-8676-28432381e6a9"}) } }),
> raw: Response { status: StatusCode(400), headers: Headers { headers:
> {"x-amzn-requestid": HeaderValue { _private: H0("1b8bbafd-5b71-4ba5-8676-28432381e6a9") },
> "x-amzn-errortype": HeaderValue { _private:
> H0("BadRequestException:http://internal.amazon.com/coral/com.amazonaws.transcribe.streaming/") },
> "date": HeaderValue { _private: H0("Tue, 26 Mar 2024 17:41:31 GMT") },
> "content-type": HeaderValue { _private: H0("application/x-amz-json-1.1") },
> "content-length": HeaderValue { _private: H0("315") }} }, body: SdkBody {
> inner: Once(Some(b"{\"Message\":\"1 validation error detected: Value 'test'
> at 'languageCode' failed to satisfy constraint: Member must satisfy enum value
> set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI, pl-PL, no-NO, nl-NL, pt-PT,
> es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN, en-AU, pt-BR, sv-SE, ja-JP,
> ca-ES, es-US, fr-CA, en-GB]\"}")), retryable: true }, extensions: Extensions {
> extensions_02x: Extensions, extensions_1x: Extensions } } })

This commit adopts the most informative and concise solution I could come up
with to log AWS errors. With the above error case, this results in:

> service error: Error { code: "BadRequestException", message: "1 validation
> error detected: Value 'test' at 'languageCode' failed to satisfy constraint:
> Member must satisfy enum value set: [ar-AE, zh-HK, en-US, ar-SA, zh-CN, fi-FI,
> pl-PL, no-NO, nl-NL, pt-PT, es-ES, th-TH, de-DE, it-IT, fr-FR, ko-KR, hi-IN,
> en-AU, pt-BR, sv-SE, ja-JP, ca-ES, es-US, fr-CA, en-GB]",
> aws_request_id: "a40a32a8-7b0b-4228-a348-f8502087a9f0" }

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1521>
2024-03-26 20:05:32 +01:00
François Laignel
9f27bde36a aws: use fixed BehaviorVersion
Quoting [`BehaviorVersion` documentation]:

> Over time, new best-practice behaviors are introduced. However, these
> behaviors might not be backwards compatible. For example, a change which
> introduces new default timeouts or a new retry-mode for all operations might
> be the ideal behavior but could break existing applications.

This commit uses `BehaviorVersion::v2023_11_09()`, which is the latest
major version at the moment. When a new major version is released, the method
will be deprecated, which will warn us of the new version and let us decide
when to upgrade, after any changes if required. This is safer that using
`latest()` which would silently use a different major version, possibly
breaking existing code.

[`BehaviorVersion` documentation]: https://docs.rs/aws-config/1.1.8/aws_config/struct.BehaviorVersion.html

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1520>
2024-03-26 17:44:16 +01:00
Sebastian Dröge
f97150aa58 reqwest: Update to reqwest 0.12
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1510>
2024-03-23 14:30:31 +02:00
Philippe Normand
be12c0a5f7 Fix clippy warnings after upgrade to Rust 1.77
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1512>
2024-03-21 17:33:32 +00:00
François Laignel
c5e7e76e4d webrtcsrc: add do-retransmission property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1509>
2024-03-21 07:25:30 +00:00
François Laignel
5476e3d759 webrtcsink: prevent video-info error log for audio streams
The following error is logged when `webrtcsink` is feeded with an audio stream:

> ERROR video-info video-info.c:540:gst_video_info_from_caps:
>       wrong name 'audio/x-raw', expected video/ or image/

This commit bypasses `VideoInfo::from_caps` for audio streams.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1511>
2024-03-20 19:57:45 +01:00
François Laignel
cc7b7d508d rtp: gccbwe: don't break downstream assumptions pushing buffer lists
Some elements in the RTP stack assume all buffers in a `gst::BufferList`
correspond to the same timestamp. See in [`rtpsession`] for instance.
This also had the effect that `rtpsession` did not create correct RTCP as it
only saw some of the SSRCs in the stream.

`rtpgccbwe` formed a packet group by gathering buffers in a `gst::BufferList`,
regardless of whether they corresponded to the same timestamp, which broke
synchronization under certain circonstances.

This commit makes `rtpgccbwe` push the buffers as they were received: one by one.

[`rtpsession`]: bc858976db/subprojects/gst-plugins-good/gst/rtpmanager/gstrtpsession.c (L2462)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1502>
2024-03-20 18:19:14 +00:00
Sebastian Dröge
cca3ebf520 rtp: Switch from chrono to time
Which allows to simplify quite a bit of code and avoids us having to
handle some API deprecations.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1503>
2024-03-20 15:05:39 +02:00
Guillaume Desmottes
96337d5234 webrtc: allow resolution and framerate input changes
Some changes do not require a WebRTC renegotiation so we can allow
those.

Fix #515

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1498>
2024-03-18 14:52:01 +01:00
Tim-Philipp Müller
eb49459937 rtp: m2pt: add some unit tests
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Tim-Philipp Müller
ce3960f37f rtp: Add MPEG-TS RTP payloader
Pushes out pending TS packets on EOS.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Tim-Philipp Müller
9f07ec35e6 rtp: Add MPEG-TS RTP depayloader
Can handle different packet sizes, also see:
https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1310

Has clock-rate=90000 as spec prescribes, see:
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/-/issues/691

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1493>
2024-03-16 10:07:37 +00:00
Guillaume Desmottes
8f997ea4e3 webrtc: janus: handle 'hangup' messages from Janus
Fix error about this message not being handled:

{
   "janus": "hangup",
   "session_id": 4758817463851315,
   "sender": 4126342934227009,
   "reason": "Close PC"
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
2024-03-13 10:14:38 +00:00
Guillaume Desmottes
992f8d9a5d webrtc: janus: handle 'destroyed' messages from Janus
Fix this error when the room is destroyed:

ERROR   webrtc-janusvr-signaller imp.rs:413:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignallerU64@0x55b166a3fe40> Unknown message from server: {
   "janus": "event",
   "session_id": 6667171862739941,
   "sender": 1964690595468240,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "destroyed",
         "room": 8320333573294267
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
2024-03-13 10:14:38 +00:00
Guillaume Desmottes
9c6a39d692 webrtc: janus: handle (stopped-)talking events
Expose those events using a signal.

Fix those errors when joining a Janus room configured with
'audiolevel_event: true'.

ERROR   webrtc-janusvr-signaller imp.rs:408:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignaller@0x560cf2a55100> Unknown message from server: {
   "janus": "event",
   "session_id": 2384862538500481,
   "sender": 1867822625190966,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "talking",
         "room": 7564250471742314,
         "id": 6815475717947398,
         "mindex": 0,
         "mid": "0",
         "audio-level-dBov-avg": 37.939998626708984
      }
   }
}
ERROR   webrtc-janusvr-signaller imp.rs:408:gstrswebrtc::janusvr_signaller:👿:Signaller::handle_msg:<GstJanusVRWebRTCSignaller@0x560cf2a55100> Unknown message from server: {
   "janus": "event",
   "session_id": 2384862538500481,
   "sender": 1867822625190966,
   "plugindata": {
      "plugin": "janus.plugin.videoroom",
      "data": {
         "videoroom": "stopped-talking",
         "room": 7564250471742314,
         "id": 6815475717947398,
         "mindex": 0,
         "mid": "0",
         "audio-level-dBov-avg": 40.400001525878906
      }
   }
}

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1481>
2024-03-13 10:14:38 +00:00
François Laignel
5b01e43a12 webrtc: update further to WebRTCSessionDescription sdp accessor changes
See: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1406
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1491>
2024-03-11 13:39:19 +01:00
Zhao, Gang
7a46377627 rtp: tests: Simplify loop
All buffers can be added in 100 outer loops. Add buffer less than 200 in the last (i = 99) loop.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1489>
2024-03-10 16:47:30 +08:00
Guillaume Desmottes
612f863ee9 webrtc: janusvrwebrtcsink: add 'use-string-ids' property
Instead of exposing all ids properties as strings, we now have two
signaller implementations exposing those properties using their actual
type. This API is more natural and save the element and application
conversions when using numerical ids (Janus's default).

I also removed the 'joined-id' property as it's actually the same id as
'feed-id'. I think it would be better to have a 'janus-state' property or
something like that for applications willing to know when the room has
been joined.
This id is also no longer generated by the element by default, as Janus
will take care of generating one if not provided.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1486>
2024-03-07 09:34:58 +01:00
Sebastian Dröge
2839e0078b rtp: Port RTP AV1 payloader/depayloader to new base classes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1472>
2024-03-06 09:40:35 +00:00
Jordan Yelloz
0414f468c6 livekit_signaller: Added missing getter for excluded-producer-peer-ids
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1484>
2024-03-04 10:08:11 -07:00
Jordan Yelloz
8b0731b5a2 webrtcsrc: Removed incorrect URIHandler from LiveKit source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1484>
2024-03-04 09:44:01 -07:00
Jordan Yelloz
002dc36ab9 livekit_signaller: Improved shutdown behavior
Without sending a Leave request to the server before disconnecting, the
disconnected client will appear present and stuck in the room for a little
while until the server removes it due to inactivity.

After this change, the disconnecting client will immediately leave the room.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1482>
2024-02-29 08:21:13 -07:00
Jordan Yelloz
f0b408d823 webrtcsrc: Removed flag setup from WhipServerSrc
It's already done in the base class

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:58 -07:00
Jordan Yelloz
17b2640237 webrtcsrc: Updated readme for LiveKit source
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:58 -07:00
Jordan Yelloz
fa006b9fc9 webrtcsrc: Added LiveKit source element
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:58 -07:00
Jordan Yelloz
96037fbcc5 webrtcsink: Updated livekitwebrtcsink for new signaller constructor
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:58 -07:00
Jordan Yelloz
730b3459f1 livekit_signaller: Added dual-role support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1461>
2024-02-28 11:25:49 -07:00
Guillaume Desmottes
60bb72ddc3 webrtc: janus: add joined-id property to the signaller
Fix #504

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 15:05:11 +01:00
Guillaume Desmottes
eabf31e6d0 webrtc: janus: rename RoomId to JanusId
Those weird ids are used in multiple places, not only for the room id,
so best to have a more generic name.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 15:05:11 +01:00
Guillaume Desmottes
550018c917 webrtc: janus: room id not optional in 'joined' message
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 14:16:46 +01:00
Guillaume Desmottes
0829898d73 webrtc: janus: remove 'audio' and 'video' from publish messages
Those are deprecated and no longer used.

See https://janus.conf.meetecho.com/docs/videoroom and
https://github.com/meetecho/janus-gateway/blob/master/src/plugins/janus_videoroom.c#L9894

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1480>
2024-02-28 13:39:04 +01:00
Guillaume Desmottes
ec17c58dee webrtc: janus: numerical room ids are u64
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1478>
2024-02-28 11:56:44 +01:00
Yorick Smilda
563eff1193 Implement GstWebRTCAPI as class instead of global instance
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1373>
2024-02-27 12:30:13 +00:00
Jordan Yelloz
594400a7f5 webrtcsrc: Made producer-peer-id optional
It may be necessary for some signalling clients but the source element
doesn't need to depend on it.

Also, the value will fall back to the pad's MSID for the first argument
to the request-encoded-filter gobject signal when it isn't available
from the signalling client.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1477>
2024-02-26 13:41:40 -07:00
Xavier Claessens
f7ffa13543 janusvr: Add string-ids property
It forces usage of strings even if it can be parsed into an integer.
This allows joining room `"133"` in a server configured with string
room ids.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1466>
2024-02-26 11:10:00 +00:00
Xavier Claessens
23955d2dbb janusvr: Room IDs can be strings
Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1466>
2024-02-26 11:10:00 +00:00
Sebastian Dröge
f563f8334b rtp: Add PCMU/PCMA RTP payloader / depayloader elements
These come with new generic RTP payloader, RTP raw-ish audio payloader
and RTP depayloader base classes.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1424>
2024-02-23 14:43:45 +02:00
Maksym Khomenko
da21dc853d webrtcsink: extensions: separate API and signal checks
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1469>
2024-02-20 19:29:46 +02:00
Maksym Khomenko
2228f882d8 webrtcsink: apply rustfmt
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1469>
2024-02-20 19:29:28 +02:00
Xavier Claessens
2572afbf15 janusvr: Add secret-key property
Every API calls have an optional "apisecret" argument.

Sponsored-by: Netflix Inc.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1465>
2024-02-16 14:04:59 +00:00
Sebastian Dröge
8ef12a72e8 rtpgccbwe: Don't reset PTS/DTS to None
The element is usually placed before `rtpsession`, and `rtpsession`
needs the PTS/DTS for correctly determining the running time. The
running time is then used to produce correct RTCP SR, and to potentially
update an NTP-64 RTP header extension if existing on the packets.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/496

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1462>
2024-02-14 08:05:54 +00:00
Jordan Yelloz
67b7cf9764 webrtcsink: Added sinkpad with "msid" property
This forwards to the webrtcbin sinkpad's msid when specified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1442>
2024-02-12 15:04:44 +00:00
Sebastian Dröge
b2d5ee48cd Update to async-tungstenite 0.25
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1455>
2024-02-11 11:31:24 +02:00
Sebastian Dröge
92891a61e8 Fix a couple of compiler/clippy warnings with --no-default-features
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1452>
2024-02-08 13:02:55 +02:00
Nirbheek Chauhan
cf5e7f6ed3 rtspsrc2: Add some top-level documentation
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
7a1cd675c2 rtspsrc2: Fix RTCP send/recv in the multicast case
Don't use connect(), since that is incompatible with multicast.
Instead, drop received packets that are from senders we do not want.

Also set multicast loopback = false so we don't receive RTCP RRs from
ourselves and interpret them as RTCP SRs.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
e59f3bbe58 rtspsrc2: Increase RTP timeout to 5 seconds, matching rtspsrc
Also fix some logging.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
3e963e9239 rtspsrc2: Implement NetAddressMeta support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-08 07:21:51 +05:30
Nirbheek Chauhan
42425abb69 rtspsrc: Factor out SDP → Caps, parse more attributes
This could be a struct of some kind derived from sdp_types::Media etc,
but this is fine for now.

Adds parsing of framesize, and fallbacks for missing or incomplete
rtpmap.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
437326ebfd rtspsrc2: Allocate a buffer pool for UDP RTP data
Control the size with a receive-mtu property

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
44e49a06a0 rtspsrc2: Emit EOS if any ssrc gets a BYE packet or times out
This allows us to exit when the live-stream ends.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
975556c06b rtspsrc2: Allow a SETUP response without a Transports header
If we only send a single Transport in the Transports header, then the
server is allowed to omit it in the response. This has some strange
consequences for UDP transport: specifically, we have no idea what
addr/port we will get the packets from.

In those cases, we connect() on the socket when we receive the first
packet, so we can send RTCP RRs, and also so we can ensure that we
ignore data from other addresses.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:23 +05:30
Nirbheek Chauhan
086ffd7aff New RTSP source plugin with live streaming support
GST_PLUGIN_FEATURE_RANK=rtspsrc2:1 gst-play-1.0 [URI]

Features:
* Live streaming N audio and N video
  - With RTCP-based A/V sync
* Lower transports: TCP, UDP, UDP-Multicast
* RTP, RTCP SR, RTCP RR
* OPTIONS DESCRIBE SETUP PLAY TEARDOWN
* Custom UDP socket management, does not use udpsrc/udpsink
* Supports both rtpbin and the rtpbin2 rust rewrite
  - Set USE_RTPBIN2=1 to use rtpbin2 (needs other MRs)
* Properties:
  - protocols selection and priority (NEW!)
  - location supports rtsp[ut]://
  - port-start instead of port-range

Co-Authored-by: Tim-Philipp Müller <tim@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1425>
2024-02-07 20:29:18 +05:30
Bilal Elmoussaoui
0615a16124 Use workspace features for crates metadata/deps
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1446>
2024-02-05 15:34:31 +01:00
Sebastian Dröge
91abc62ad0 webrtcsink: Fix new clippy warning
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:53:20 +02:00
Sebastian Dröge
1a55c70114 Switch git dependencies to explicitly name branch
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1445>
2024-02-05 12:51:36 +02:00
Sebastian Dröge
ffa830ae9b Update for GLib prelude re-organization
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1444>
2024-02-03 12:30:15 +02:00
Jordan Yelloz
311fda649f livekit_signaller: Added high-quality layer for video streams
This change addresses a cosmetic issue with livekit, where the
connection quality indicator seen by other users shows bad quality
unless the track is added with a high quality layer. The details of the
layer submitted aren't significant for this purpose.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1443>
2024-02-02 20:57:17 +00:00
Robert Ayrapetyan
916a8b959e doc: add http headers for webrtcsink signaller
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
972b9e5474 doc: add docstrings for signaller object
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
Robert Ayrapetyan
7a72b2fc25 webrtcsink-signalling: add headers support
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1419>
2024-02-01 19:31:58 +00:00
François Laignel
91bfd0f7c3 webrtc: signallers: attempt to close the ws when an error occurs
This commit discards the early error returns in the send tasks to log the error
and attempt to close the websocket.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
François Laignel
f54d714afd webrtc: only use close() to close websockets
In the signaller clients and servers, the following sequence is used to close
the websocket (in the [send task]):

```rust
    ws_sink.send(WsMessage::Close(None)).await?;
    ws_sink.close().await?;
```

tungstenite's [`WebSocket::close()` doc] states:

> Calling this function is the same as calling `write(Message::Close(..))``

So we might think they are redundant and either could be used for this purpose
(`send()` calls `write()`, then `flush()`).

The result is actually is bit different as `write()` starts by checking the
state of the connection and [returns `SendAfterClosing`] if the socket is no
longer active, which is the case when a closing request has been received from
the peer via a [call to `do_close()`]). Note that `do_close()` also enqueues a
`Close` frame.

This behaviour is visible from the server's logs:

```
1. tungstenite::protocol: Received close frame: None
2. tungstenite::protocol: Replying to close with Frame { header: FrameHeader { .., opcode: Control(Close), .. }, payload: [] }
3. gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
4. gst_plugin_webrtc_signalling::server: connection closed: None this_id=cb13892f-b4d5-4d59-95e2-b3873a7bd319
5. remove_peer{peer_id="cb13892f-b4d5-4d59-95e2-b3873a7bd319"}: gst_plugin_webrtc_signalling::server: close time.busy=285µs time.idle=55.5µs
6. async_tungstenite: websocket start_send error: WebSocket protocol error: Sending after closing is not allowed
```

1: The server's websocket receives the peer's `Close(None)`.
2: `do_close()` enqueues a `Close` frame.
3: The incoming `Close(None)` is handled by the server.
4 & 5: perform session closing.
6: `ws_sink.send(WsMessage::Close(None))` attempts to `write()` while the ws
   is no longer active. The error causes an early return, which means that
   the enqueued `Close` frame is not flushed.

Depending on the peer's shutdown sequence, this can result in the following
error, which can bubble up as a `Message` on the application's bus:

```
ERROR: from element /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0: GStreamer encountered a general stream error.
Additional debug info:
net/webrtc/src/webrtcsrc/imp.rs(625): gstrswebrtc::webrtcsrc:👿:BaseWebRTCSrc::connect_signaller::{{closure}}::{{closure}} (): /GstPipeline:pipeline0/GstWebRTCSrc:webrtcsrc0:
Signalling error: Error receiving: WebSocket protocol error: Connection reset without closing handshake
```

On the other hand, [`close()` ensures the ws is active] before attempting to
write a `Close` frame. If it's not, it only flushes the stream.

Thus, when we want to be able to close the websocket and/or to honor the closing
handshake in response to the peer `Close` message, the `ws_sink.close()`
variant is preferable.

This can be verified in the resulting server's logs:

```
tungstenite::protocol: Received close frame: None
tungstenite::protocol: Replying to close with Frame { header: FrameHeader { is_final: true, rsv1: false, rsv2: false, rsv3: false, opcode: Control(Close), mask: None}, payload: [] }
gst_plugin_webrtc_signalling::server: Received message Ok(Close(None))
gst_plugin_webrtc_signalling::server: connection closed: None this_id=192ed7ff-3b9d-45c5-be66-872cbe67d190
remove_peer{peer_id="192ed7ff-3b9d-45c5-be66-872cbe67d190"}: gst_plugin_webrtc_signalling::server: close time.busy=22.7µs time.idle=37.4µs
tungstenite::protocol: Sending pong/close
```

We now get the notification `Sending pong/close` (the closing handshake) instead
of `websocket start_send error` from step 6 with previous variant.

The `Connection reset without closing handshake` was not observed after this
change.

[send task]: 63b568f4a0/net/webrtc/signalling/src/server/mod.rs (L165)
[`WebSocket::close()` doc]: https://docs.rs/tungstenite/0.21.0/tungstenite/protocol/struct.WebSocket.html#method.close
[returns `SendAfterClosing`]: 85463b264e/src/protocol/mod.rs (L437)
[call to `do_close()`]: 85463b264e/src/protocol/mod.rs (L601)
[`close()` ensures the ws is active]: 85463b264e/src/protocol/mod.rs (L531)

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1435>
2024-02-01 18:08:41 +01:00
Taruntej Kanakamalla
50e905fe4b webrtc: conditional compile for features with 1_22 dependency
Few features being used in webrtcsink like
the signal `request-aux-sender` are introduced
to webrtcbin in gstreamer release 1.22.

Rename the feature gst1_22 to v1_22 for uniformity.

Add v1_22 to default features.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1241>
2024-02-01 15:08:11 +05:30
Sebastian Dröge
f2a7a34abf rtp: gcc: Use x += ... instead of x = x + ... 2024-01-31 18:46:55 +02:00
Sebastian Dröge
4ad101b53b Use once_cell crate directly again
The glib crate does not depend on it anymore and also does not re-export
it anymore.

Also switch some usages of OnceCell to OnceLock from std.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sebastian Dröge
451d928026 webrtc: Update AWS signaller to http 1
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1441>
2024-01-31 18:07:57 +02:00
Sanchayan Maity
95c007953c webrtchttp: Allow audio or video caps to be specified as None with WHEP
We were setting audio and video caps by default even when the user
might have requested only video or audio. This would then result
in a `Could not reuse transceiver` error from the webrtcbin.

Fix this by allowing the user to specify audio or video caps as
None. This allows us to maintain the earlier behaviour for backward
compatibility while allowing the user to not request audio or video
as need be.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1433>
2024-01-18 15:43:19 +05:30
Sebastian Dröge
764143d971 webrtc: Remove unnecessary manual Send+Sync implementations for signallers
These are automatically implemented.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/483

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:25 +02:00
Sebastian Dröge
1af18f3028 webrtc: Require Send+Sync for signaller implementations
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1432>
2024-01-18 10:01:01 +02:00
Eva Pace
80b58f3b45 net/webrtc/janusvr: add JanusVRWebRTCSink plugin/signaller
The JanusVRWebRTCSink is a new plugin that integrates with the Video
Room plugin of the Janus Gateway, which simplifies WebRTC communication.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1362>
2024-01-17 20:33:57 +00:00
Maksym Khomenko
773ebc7854 webrtcsrc: don't restrict RTP extensions to TWCC only
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1381>
2024-01-17 07:34:01 +00:00
Sebastian Dröge
dfa95d8ed3 webrtc: Update to livekit-api / livekit-protocol 0.3
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1427>
2024-01-16 07:52:48 +00:00
Maksym Khomenko
fecbe01e06 webrtcsink: make 'extensions' property usage conditional
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1423>
2024-01-16 07:13:56 +00:00
Sebastian Dröge
73a53e38c4 aws: s3: Disable remaining tests too for now
They fail state changes, which cases `GstHarness` to abort.
2024-01-16 09:13:41 +02:00
Arun Raghavan
fd3675aac0 aws: s3: Temporarily disable putobject tests
Disabling while we figure out why it's failing.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1429>
2024-01-15 21:43:25 -05:00
Arun Raghavan
8b18ca15b5 Revert "aws: Disable putobjectsink tests for now"
This reverts commit b128d127c2.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11 15:38:36 -05:00
Arun Raghavan
06213714c5 aws: putobjectsink: Fix a couple of minor log typos
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1416>
2024-01-11 15:38:36 -05:00
Nirbheek Chauhan
2d85048925 webrtc/signalling: We get the address when accepting
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Nirbheek Chauhan
63b568f4a0 webrtc/signalling: Fix potential hang and FD leak
If a peer connects via TCP and never initiates TLS, then the server
will get stuck in the accept loop. Spawn a task when accepting a TLS
connection, and timeout if it doesn't complete in 5 seconds.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1412>
2023-12-29 13:28:48 +00:00
Maksym Khomenko
17f0b61576 webrtcsink: add payloader-setup signal
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1389>
2023-12-23 08:02:08 +00:00
Sebastian Dröge
b128d127c2 aws: Disable putobjectsink tests for now
See https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/472

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1413>
2023-12-22 13:25:12 +02:00
Arun Raghavan
6d47045a60 aws: s3sink: Fix spelling of debug category
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Arun Raghavan
410d104ad6 aws: s3putobjectsink: Add a flush-on-error property
Makes sure we can send out data even if the pipeline shutdown in error.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Arun Raghavan
12dbf50ddc aws: s3putobjectsink: Add some thresholds for flushing
Lets us connect when we perform a flush

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 16:13:48 -05:00
Arun Raghavan
a54b2dd39e aws: s3: Add a new awss3putobjectsink
When streaming small amounts of data, using awss3sink might not be a
good idea, as we need to accumulate at least 5 MB of data for a
multipart upload (or we flush on EOS).

The alternative, while inefficient, is to do a complete PutObject of
_all_ the data periodically so as to not lose data in case of a pipeline
failure. This element makes a start on this idea by doing a PutObject
for every buffer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1337>
2023-12-18 10:39:23 -05:00
Sebastian Dröge
81dd45c814 webrtc: Downgrade aws-smithy-http to 0.60
Version 0.61 was yanked from crates.io.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1407>
2023-12-14 09:11:07 +02:00
Sebastian Dröge
2f2bf6ca8f webrtc: Update to aws-smithy-http 0.61
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:21:38 +02:00
Sebastian Dröge
0bae18fe0d rtp: Update to bitstream-io 2.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:17:51 +02:00
Sebastian Dröge
181bd13103 Update to async-tungstenite 0.24
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1404>
2023-12-09 12:17:11 +02:00
Guillaume Desmottes
6dfd1c1496 use new debug and parse API
Changes from https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1355

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1403>
2023-12-04 15:58:21 +01:00
Sebastian Dröge
f13574d8ed Update further AWS SDK crates to 1.0
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1400>
2023-11-26 10:26:02 +02:00
Mathieu Duponchelle
cf1c7600a2 webrtcsink: don't panic on failure to request pad from webrtcbin
webrtcbin will refuse pad requests for all sorts of reasons, and should
be logging an error when doing so, simply post an error message and let
the application deal with it, the reason for the refusal should
hopefully be available in the logs to the user.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1399>
2023-11-24 19:53:38 +01:00
Sebastian Dröge
c3ced8c7e6 Update to AWS SDK 1.0 / 0.60 / 0.39
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1397>
2023-11-21 10:32:59 +02:00
Sebastian Dröge
1d9c89e3fe Update to AWS SDK 0.101 / 0.59 / 0.38 2023-11-20 10:13:13 +02:00
Sebastian Dröge
66c62d69b9 aws: Stop using deprecated aws_config function in the test 2023-11-18 10:16:24 +02:00
Taruntej Kanakamalla
43ee6bfc1c net/webrtc: add whipserversrc
Implement new signaller WhipServerSignaller
 - an http server using 'warp'
 - handlers for the POST, OPTIONS, PATCH and DELETE
 - fixed path `/whip/endpoint` as the URI
 - fixed value 'whip-client' as the producer peer id
 - fixed resource url `/whip/resource/whip-client`

Derive whipserversrc element from BaseWebRTCSrc
 - implement constructed method for ObjectImpl to set
  non-default signaller, i.e., WhipServerSignaller
 - bind the properties stun-server and turn-servers to those on
   the Signaller

Connect to 'webrtcbin-ready' signal in the constructor of WhipServerSignaller
 - it will be emitted by the webrtcsrc when the webrtcbin element is ready
 - the closure for this signal will in turn connect to webrtcbin's ice-gathering-state
   and perform send with the answer sdp via the channel
 - the WhipServer will hold its HTTP response in POST handler until this signal
   is received or timeout which happens early

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
ed3aa740be net/webrtc: deprecate consumer-added on the signaller
add a new signal webrtcbin-ready in this place doing same
thing but can be used for both consumers and producers

Please note this change is only to the consumer-added
signal on the signaller interface.
The consumer-added signal on the webrtcsink is unchanged

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
2d3d03b4d3 net/webrtc: rename WhipSignaller as WhipClientSignaller
remove generalized names to accommodate for the WhipServer
- name the Signaller for whipsink as WhipClient
- name the Settings for whipsink as WhipClientSettings
- name the State for whipsink as WhipClientState

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Taruntej Kanakamalla
a0638ec983 net/webrtc: Extract BaseWebRTCSrc
Define a Base for all the webrtcsrc type elements
so they can all be derived from it. Similar to base
element defined for webrtcsink type elements

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1284>
2023-11-17 18:08:44 +00:00
Sebastian Dröge
dee27e35b7 Update to latest AWS SDK
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1395>
2023-11-17 11:22:29 +02:00
Sebastian Dröge
58723f2a8c Update to AWS SDK 0.36
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1394>
2023-11-15 17:20:58 +02:00
François Laignel
9250c592a7 ndi: don't accumulate meta with audio only streams
Currently, only closed caption metadata are supported. When the next video
frame is received, pending meta are dequeued and parsed. If close captions
are found, they are attached to the video frame.

For audio only streams, it doesn't make sense to enqueue metadata. They would
accumulate in `pending_metadata` and would never be dequeued.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/460

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1392>
2023-11-13 19:26:23 +01:00
Sebastian Dröge
39155ef81c ndisrc: Implement zerocopy handling for the received frames if possible
Also move processing from the capture thread to the streaming thread.
The NDI SDK can cause frame drops if not reading fast enough from it.

All frame processing is now handled inside the ndisrcdemux.

Also use a buffer pool for video if copying is necessary.

Additionally, make sure to use different stream ids in the stream-start
event for the audio and video pad.

This plugin now requires GStreamer 1.16 or newer.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 13:22:48 +02:00
Sebastian Dröge
2afffb39dd ndi: Don't mark private type as public
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Sebastian Dröge
99d7cce0d6 ndi: Refactor frame structs to have static lifetimes
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Sebastian Dröge
eb137ec6dc ndi: Remove wrong Clone impl on RecvInstance
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1365>
2023-11-13 10:29:25 +02:00
Arun Raghavan
771741c10c Revert "s3: tests: Remove emoji-based tests for now"
This reverts commit a49a5dcb11.

Now that hotdoc should work with emoji, let's bring the tests back.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1386>
2023-11-09 11:50:53 -05:00
Maksym Khomenko
e5fd2c3568 webrtcsrc: add turn-servers property
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1380>
2023-11-04 10:19:45 +00:00
Mathieu Duponchelle
5371eb52ad Port to AWS SDK 0.57/0.35
Co-authored-by: Sebastian Dröge <sebastian@centricular.com>
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1379>
2023-11-03 15:13:45 +00:00