Commit graph

3514 commits

Author SHA1 Message Date
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
Seungha Yang
ebdcc403cf transcriberbin: Fix mux-method=cea708
* Update "translation-languages" property to include G_PARAM_CONSTRUCT
so that it can be applied to initial state.

* Change default "translation-languages" value to be None instead of
cea608 specific one. Transcriberbin will be able to configure initia
state depending on selected mux method if "translation-languages" is
unspecified.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1589>
2024-05-30 04:40:09 +09:00
Matthew Waters
45800d7636 tttocea708: ensure periodic sync points in roll up mode
Otherwise, without the relevant DefineWindow, then a receiver cannot
begin to display the captions from the middle of a stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1591>
2024-05-29 11:15:10 +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
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
Sebastian Dröge
28bd6f07a2 Update CHANGELOG.md for 0.12.6 2024-05-23 17:27:21 +03:00
Sebastian Dröge
b1ad123595 gtk4: Fix Python example in the non-GL code path 2024-05-23 16:15:52 +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
b12da2c543 deny: Update with itertool 0.12 override
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1579>
2024-05-20 14:36:39 +03:00
Sebastian Dröge
02cd2c42fd Update Cargo.lock
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1579>
2024-05-20 14:33:23 +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