Setting different number of temporal layers is supported by SimulcastEncodeAdapter and LibvpxVp8Encoder will fallback to SimulcastEncoderAdapter if InitEncode fails.
Bug: none
Change-Id: I8a09ee1e6c70a0006317957c0802d019a0d28ca2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228642
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34785}
Using usrsctp_getladdrs() would sometimes be flagged by TSAN for a lock
order inversion. It was used to retrieve the "id" of the socket on the
transport.
The "id" is instead stored in the "ulp_info" parameter, which is
passed with each callback from usrsctp.
Bug: webrtc:12823
Change-Id: Ifb3d7780273a460e677526dd3a93f9365b29300c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229000
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34784}
turning the current field trial into a killswitch.
Note that RED is not used by default since it is listed after opus in the SDP.
To enable RED for opus the setCodecPreferences can be used to change
the order of codecs.
BUG=webrtc:11640
Change-Id: I248f4340ca0a3f7c4ea6d6a41b566bc92ab6f19d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228426
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34781}
Before this CL PeerConnectionE2EQualityTestSmokeTest was actually
overwriting this field trial:
[ RUN ] PeerConnectionE2EQualityTestSmokeTest.Smoke
(field_trial.cc:140): Setting field trial string:WebRTC-UseStandardBytesStats/Enabled/
(field_trial.cc:140): Setting field trial string:
(network_emulation.cc:480): Created emulated endpoint 192.168.0.0 (); id=1
(network_emulation.cc:480): Created emulated endpoint 192.168.0.1 (); id=2
(field_trial.cc:140): Setting field trial string:WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/
After this CL it is instead used:
[ RUN ] PeerConnectionE2EQualityTestSmokeTest.Smoke
(network_emulation.cc:480): Created emulated endpoint 192.168.0.0 (); id=1
(network_emulation.cc:480): Created emulated endpoint 192.168.0.1 (); id=2
(field_trial.cc:140): Setting field trial string:WebRTC-UseStandardBytesStats/Enabled/WebRTC-FlexFEC-03-Advertised/Enabled/WebRTC-FlexFEC-03/Enabled/
This CL also removes the non effective field trial override in
test/pc/e2e/peer_connection_e2e_smoke_test.cc which was unset as soon
as the variable was going out of scope.
Bug: b/186198412
Change-Id: I1698407e2c490a80c1f835cd591624446cf993fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229023
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34779}
The congestion window is unlikely to be even divisible by the size
of a packet, so when the congestion window is almost full, there is
often just a few bytes remaining in it. Before this change, a small
packet was created to fill the remaining bytes in the congestion window,
to make it really full.
Small packets don't add much. The cost of sending a small packet is
often the same as sending a large one, and you usually get lower
throughput sending many small packets compared to few larger ones.'
This mode will only be enabled when the congestion window is large, so
if the congestion window is small - e.g. due to poor network conditions,
it will allow packets to become fragmented into small parts, in order to
fully utilize the congestion window.
Bug: webrtc:12943
Change-Id: I8522459174bc72df569edd57f5cc4a494a4b93a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228526
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34778}
For symmetry, as the outstanding_bytes is increased/decreased by
the serialized chunk size (not just the payload) - which is compared
to the congestion window, the congestion window should be increased
by the serialized size of chunks acked - not just their payload.
Bug: webrtc:12943
Change-Id: I0a06033e8ca0d58433138df6442ca80494918cf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228525
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34775}
The same code was done for both acking chunks due to moving the
cum-ack-tsn and when acking gap-ack-blocks. Unify them completely
to have a single code path.
Bug: webrtc:12943
Change-Id: I3b864e41cc2ec674460517660c23b72a70edf720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228521
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34773}
This is mainly a refactoring commit, to break out packet sending to a
dedicated component.
Bug: webrtc:12943
Change-Id: I78f18933776518caf49737d3952bda97f19ef335
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228565
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34772}
Before this change, there was no way for a client to indicate to the
dcSCTP library if a packet that was supposed to be sent, was actually
sent. It was assumed that it always was.
To handle temporary failures better, such as retrying to send packets
that failed to be sent when the send buffer was full, this information
is propagated to the library.
Note that this change only covers the API and adaptations to clients.
The actual implementation to make use of this information is done as a
follow-up change.
Bug: webrtc:12943
Change-Id: I8f9c62e17f1de1566fa6b0f13a57a3db9f4e7684
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228563
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34767}
We already default to PipeWire 0.3 and there is no reason to keep
continue supporting an old version of PipeWire which is not maintained
anymore, wont't get any update or new features. It also makes the code
easier to understand since we can remove all ifdefs we had to support
two versions simultaneously.
Bug: chromium:1146942
Change-Id: I7156e1784ebfad111485a2944199563568a75eec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227345
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34765}
Enabling bitcode doesn't seem to create a separate dSYM.
To make it work in this configuration, when creating an XCFramework,
pass dSYM only if it exists.
Bug: none
Change-Id: I6d95dc765accc10a70caeb88063d05eeea630dd1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34762}
from runtime check in proxy classes that picks decoder (VCMDecoderDataBase)
to a DCHECK in the VideoDecoder::Settings
Bug: None
Change-Id: Ic8c2e971486a3a7eb247f9d03815aba5ca5a7bad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228644
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34761}
Task posted by OnSendRtp might be scheduled after `send_stream_` is
destroyed. Fix by using a PendingTaskSafetyFlag, killed from the
OnStreamsStopped callback.
Bug: webrtc:12726
Change-Id: I935917a3d80e82c3536261d72059448fb7aac00d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228643
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34754}
As a first step we only want to enable frame pacing for the case
where min playout delay == 0 and max playout delay > 0.
Bug: chromium:1237402, chromium:1239469
Change-Id: Icf9641db7566083d0279135efa8618e435d881eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228640
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34752}
The UDP sockets in WebRTC are non-blocking, and when writing too much
to them so that their send buffer becomes exhausted, they will return
EAGAIN or EWOULDBLOCK, which indicates that the client will need to
retry a bit later.
Media packets are generally sent by the pacer, which generally avoids
this exhaustion, but for SCTP which has a congestion control algorithm
quite similar to TCP, it may overshoot the amount of data it writes. If
the SCTP library can be notified when writing fails, it can stop writing
for a while until the socket recovers, which will result in less
overshooting and fewer lost packets (possibly even none). But for the
SCTP library to be able to know this, errors must be propagated, which
they weren't with the argument that packets may get lost anyway.
Bug: webrtc:12943
Change-Id: I9244580abf0d48ff810da30a23f995d12be623ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228439
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34751}
Recently CPD team rolled out upload completion token feature to all users. Pressure on the system increased. Now became more common situations when upload completed, but because of Datastore limitations we can't see confirmation of it for some measurements.
I've checked 6 recent failures. For all of them amount of timeout measurements were less than 3% (less than 15 in absolute numbers, the biggest percent of failures was for 80 measurements, 2 of which timed out).
Bug: b/182111579
Change-Id: Ia5af367870d1cf7d28b9422c4114c6b85c41f865
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228562
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34749}
Some deployments, e.g. Chromium, has a limited send buffer. It's
reasonable that it's quite small, as it avoids queuing too much, which
typically results in increased latency for real-time communication. To
avoid SCTP to fill up the entire buffer at once - especially when doing
fast retransmissions - limit the amount of packets that are sent in one
go.
In a typical scenario, SCTP will not send more than three packets for
each incoming packet, which is is the case when a SACK is received which
has acknowledged two large packets, and which also adds the MTU to the
congestion window (due to in slow-start mode), which then may result in
sending three packets. So setting this value to four makes any
retransmission not use that much more of the send buffer.
This is analogous to usrsctp_sysctl_set_sctp_fr_max_burst_default in
usrsctp, which also has the default value of four (4).
Bug: webrtc:12943
Change-Id: Iff76a1668beadc8776fab10312ef9ee26f24e442
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228480
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34744}
This is a private method, and the `prepare_address` argument was
constant true at all call sites.
Bug: None
Change-Id: Id4714ee7c154a729a6e106c29895fc760e9d0455
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228527
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34739}
Currently, the RtpRtcp module of AudioSendStream is (de)registered in
the packet router on calls to
(Register|Reset)SenderCongestionControlObjects.
This CL changes that to happen on Start/Stop instead, which allows us
to safely call (Get|Set)RtpState on suspend/resume without the need
for extra locking in the rtp module.
See also https://webrtc-review.googlesource.com/c/src/+/228430
Bug: webrtc:11340
Change-Id: I54243a9ace8a7659924269418468b49b967b9465
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/228433
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34738}
