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Author SHA1 Message Date
Sebastian Dröge
ac79b52cff 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/1595>
 2024-06-13 12:37:52 +00:00
Martin Nordholts
d0dc85293d 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/1582>
 2024-05-23 16:18:01 +03:00
Martin Nordholts
be72988545 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/1582>
 2024-05-23 16:17:56 +03:00
Martin Nordholts
536601e65d 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/1582>
 2024-05-23 16:14:20 +03:00
Martin Nordholts
26f63c5e1e rtpgccbwe: Add mising 'ps' suffix to 'kbps' of 'effective bitrate' 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1582>
 2024-05-23 16:14:13 +03:00
Martin Nordholts
3a6c663f3c 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/1582>
 2024-05-23 16:13:53 +03:00
Sebastian Dröge
b498c44df5 rtpgccbwe: Move away from deprecated time::Instant to std::time::Instant 
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1556>
 2024-04-29 11:50:29 +03:00
François Laignel
47429e2ed8 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/1556>
 2024-04-29 11:49:21 +03:00
Martin Nordholts
0faa74b74d 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/1531>
 2024-04-08 15:18:03 +03:00
François Laignel
158fe80779 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/1508>
 2024-03-21 13:30:20 +02:00
Sebastian Dröge
c982db73a7 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/1508>
 2024-03-21 13:30:20 +02:00
Sebastian Dröge
3679db5740 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/1476>
 2024-02-26 14:20:53 +02:00
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