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141 commits

Author SHA1 Message Date
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
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
16b917abb1 Update for gst::Rank API changes 2023-11-02 14:10:59 +02:00
Sebastian Dröge
829469d0fe rtpav1depay: Don't push stale temporal delimiters downstream 
Only push them downstream once a complete OBU was assembled.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
 2023-10-24 11:13:35 +00:00
Sebastian Dröge
1f5e9a9335 rtpav1depay: Skip unexpected leading fragments 
If a packet is starting with a leading fragment but we do not expect to
receive one, then skip over it to the next OBU.

Not doing so would cause parsing of the middle of an OBU, which would
most likely fail and cause unnecessary warning messages about a
corrupted stream.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1367>
 2023-10-24 11:13:35 +00:00
Sebastian Dröge
d688aeb184 Update versions to 0.12.0-alpha.1 2023-08-10 17:21:11 +03:00
Sebastian Dröge
3b41f206bc Don't generate .def files for plugins 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/389

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1299>
 2023-08-09 13:54:34 +03:00
Sebastian Dröge
31b1cb8ca6 Update minimum supported Rust version to 1.70 
gtk-rs will update soonish too.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1280>
 2023-07-19 09:19:34 +03:00
Bilal Elmoussaoui
dd2d7d9215 Use re-exported once_cell 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1268>
 2023-07-06 17:50:49 +03:00
Sebastian Dröge
c65b3429ad Use MPL as license specifier for plugins only requiring GStreamer < 1.20 
And use MPL-2.0 for all that require GStreamer 1.20 or newer. The new
string is only allowed in 1.20 or newer and using it in older versions
causes failure to load the plugin.

All affected plugins are of course still MPL-2.0 licensed.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/374

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1235>
 2023-06-07 19:13:55 +03:00
Edward Hervey
31b06e52ea rtpgccbwe: Improve packet handling 
Both the delay-based *and* loss-based estimates should be computed instead of
just one. This ensures faster adaptation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1179>
 2023-05-29 08:20:36 +00:00
François Laignel
7ba0073052 use Pad builders for optional name definition 
Also, apply auto-naming in the following cases

* When building from a non wildcard-named template, the name of the template is
  automatically assigned to the Pad. User can override with a specific name by
  calling `name()` on the `PadBuilder`.
* When building with a target and no name was provided via the above, the
  GhostPad is named after the target.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/issues/448
Auto-naming discussion: https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1255#note_1891181

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1197>
 2023-05-12 12:55:31 +02:00
Edward Hervey
721d17e181 rtpgccbwe: Don't process empty lists 
The structure parsing could result in an empty vector. Don't do any processing
since the loss code assumes it's non-empty for average estimates which would
result in weird/invalid results.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1181>
 2023-04-15 19:35:27 +02:00
Guillaume Desmottes
403004a85e fix typos 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1170>
 2023-04-10 13:35:32 +02:00
David Revay
002a70a2a4 chore(webrtcsink): fix max-bitrate blurb and nick 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1150>
 2023-03-28 16:11:05 +11:00
Sebastian Dröge
9fc1404415 Update minimum supported Rust version to 1.66 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1096>
 2023-02-20 11:09:01 +02:00
Seungha Yang
6420fe43da rtpav1pay: Fix Leb128Bytes size parsing 
There are multiple ways of encoding the value, and don't assume
that bitstream used the way used in this plugin. Instead, count
the number of used bytes.

Fixes: https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/312
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1090>
 2023-02-10 18:47:52 +00:00
Sebastian Dröge
1e13dbb99c Update versions to 0.11.0-alpha.1 2023-02-10 00:23:56 +02:00
Sebastian Dröge
0ed74d0aa4 rtpgccbwe: Don't use clamp() if there's no clear min/max value 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/305

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1078>
 2023-02-06 21:56:46 +02:00
Sebastian Dröge
5506f8001e rtpav1pay: Add support for tu/frame aligned input 
In this case every buffer can be sent out immediately and makes up a
whole frame.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge
194c4e9e9f rtpav1pay: Consider the marker flag to output packets immediately at the end of a frame 
Otherwise it is necessary to wait for the beginning of the following
frame, which unnecessarily increases the latency.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/255

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge
49350f738f rtpav1depay: Fix depayloading of packets starting with a leading OBU fragment followed by more OBUs 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/288

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge
1756d7a516 rtpav1depay: Fix error handling 
Don't error out immediately on errors anymore but try again with the
next packet.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/289

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge
ed4e9a50d5 rtpav1depay: Set DISCONT flag on buffers following a corrupted packet 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge
d6cb9d72d8 rtpav1depay: Don't output full TUs but just OBUs as they come 
Simplifies state tracking and potentially reduces latency as it's not
necessary to wait until all fragments of an OBU are received.

The last OBU of a TU is marked with the marker flag to allow parsers to
detect this without first seeing the beginning of the next TU.

Also use a simple `Vec` for collecting complete OBUs instead of a
`gst_base::Adapter` as this reduces the number of allocations.

And also handle invalid packets a little bit more gracefully.

Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/244

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1072>
 2023-02-02 20:24:27 +02:00
Sebastian Dröge
560bdc4cb7 Update for glib API changes 2023-01-31 12:24:07 +02:00
Sebastian Dröge
3b4c48d9f5 Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1062>
 2023-01-25 10:31:19 +02:00
Sebastian Dröge
2c386fb792 Update for various deprecated APIs 2023-01-22 20:07:26 +02:00
Sebastian Dröge
4582ae91ab Move remaining plugins to ParamSpec builders 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1054>
 2023-01-21 18:34:55 +02:00
Philippe Normand
0fd63ece7d rtpav1depay: Implement srcpad set_caps 
Without this auto-pluggers such as decodebin or parsebin will be unable to
process AV1 RTP payloads.

Tested with: `videotestsrc num-buffers=50 ! videoconvert ! av1enc ! av1parse ! rtpav1pay ! queue ! decodebin3 ! videoconvert ! queue ! autovideosink`

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1034>
 2023-01-03 19:35:45 +02:00
Sebastian Dröge
3f904553ea Fix various new clippy warnings 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1011>
 2022-12-13 11:43:16 +02:00
Sebastian Dröge
a8250abbf1 Fix various new clippy warnings 2022-11-01 10:27:48 +02:00
Sebastian Dröge
f2223cf2cb Update versions to 0.10.0-alpha.1 2022-10-24 19:31:19 +03:00
Sebastian Dröge
9a68f6e221 Move from imp.instance() to imp.obj() 
It's doing the same thing and is shorter.
 2022-10-23 23:08:46 +03:00
François Laignel
86776be58c Remove & for obj in log macros 
This is no longer necessary.

See https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/-/merge_requests/1137
 2022-10-23 21:22:31 +02:00
Sebastian Dröge
f045099fc1 Fix GObject type names, GStreamer debug category names and element factory names 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/198
 2022-10-23 20:46:08 +03:00
Sebastian Dröge
5d44e0eb3c rtp: Move GCC bandwidth estimation element from webrtc to rtp plugin 2022-10-23 20:25:08 +03:00
Sebastian Dröge
45168639e9 Rename rtpav1 plugin to just rtp 
Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/issues/243
 2022-10-23 20:04:43 +03:00