This would allow network controllers, GoogCcNetworkController in particular to have access to Environment at construction and thus it can rely on propagated field trials and won't need to fallback to the global field trial string
Bug: webrtc:42220378
Change-Id: I36099522e3866a89a8c8d6303da03f7d5b1cad8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/350201
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42260}
This adds a new mode to the ProbeController that sends probes every
second the first 5seconds. Intented usage is before normal media has
started flowing. If enabled, the max allocated bitrat is ignored during
these initial probes.
The purpose is to have a more accurate estimate at the beginning of a
call.
The cl also removes ProbeController::SetFirstProbeToMaxBitrate.
Bug: webrtc:14928
Change-Id: I04feefb2f1b82ff38b35ee2be810f9119c53536a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349924
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42252}
in an attempt to break up the monolithic ssl target.
BUG=None
Change-Id: I38f5b3e2828742d5d918460db1af0a5797d6a5c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349764
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@meta.com>
Cr-Commit-Position: refs/heads/main@{#42249}
This reverts commit 501c4f37bf.
Patch set 1 contains pure reland.
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
Bug: webrtc:14928
Change-Id: I6a8660da20ac54237f04a29461e03b31bd988bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347643
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@google.com>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42086}
This reverts commit 33cc83595a.
Reason for revert: Perf bots showed that this cl cause a change in metrics. It looks like it is for the better, but we want this to be behind a field trial.
Original change's description:
> Ignore allocated bitrate during initial exponential BWE.
>
> The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
> That is the, initial probe will try to probe up to the max configured bitrate.
>
> ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
> continue up to the max configured bitrate, regardless of of the max
> allocated bitrate.
>
> Bug: webrtc:14928
> Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#42049}
Bug: webrtc:14928
Change-Id: I56ba58560b6857b6069552c02df822691f7af64d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347622
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Owners-Override: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42081}
The reason why we want to do this is because audio can allocate a needed bitrate before video when starting a call, which may lead to a race between the first probe result and updating the allocated bitrate.
That is the, initial probe will try to probe up to the max configured bitrate.
ProbeController::SetFirstProbeToMaxBitrate will allow the first probe to
continue up to the max configured bitrate, regardless of of the max
allocated bitrate.
Bug: webrtc:14928
Change-Id: I6e0ae90e21a78466527f3464951e6033dc846470
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346760
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42049}
It is a public interface and must be visible to allow tests to include the header file.
Bug: none
Change-Id: I4e6322c622f62c018b274b751e2c395eed7816e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/346520
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42027}
This code was extracted to make the next following CL easier to review.
This CL simply exposes the getters, setters and callbacks to set the
buffered amount low threshold on a specific SCTP stream. It will be
used in a follow-up CL, but is just boilerplate.
Bug: chromium:40072842
Change-Id: Iccd72208b369ddc252cc5886f6446b9c2ceeb0b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343360
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41943}
Before this change, calling buffered_amount only included what was
buffered on top of what was already buffered in the SCTP socket. With
the defaults, the SCTP socket can buffer up to 2MB of data (that is not
put on the wire) before the additional external bufferering in
SctpDataChannel will be used. The buffering that I am working on
removing completely.
Until it's removed completely, to avoid the issue reported in
crbug.com/41221056, include the bytes buffered in the SCTP socket to
what is returned when calling RTCDataChannel::buffered_amount.
This means that when this value is zero, it can be safe to know that all
bytes have been sent, but not necessarily acknowledged. And calling
close will not discard any messages.
This is a stopgap solution, but as functional as the proper solution
that removes all additional buffering. Follow-up CLs will merely improve
this solution.
Bug: chromium:41221056
Change-Id: I06edd52188d3bf13a17827381a15a4730722685a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/342520
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41898}
The DcSctpTransport will soon use field trials to conditionally enable
some options.
And overall, there is a migration project to start using the Environment
and this CL is in that direction, also setting the boundary; The dcSCTP
library should not depend on it. But the transport is allowed to.
Bug: webrtc:14997
Change-Id: I1f3c2c0d8dd7bdc698dd1d58bde7651b682bcba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/341480
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41872}
Using the Api, BWE components are recreated and new settings can be
applied. Initially, the only configuration available is allowing BWE probes without media".
Note that BWE components are created when transport first becomes writable. So calling this method before a PeerConnection is connected is cheap and only changes configuration.
Integration test in https://webrtc-review.googlesource.com/c/src/+/337322
Bug: webrtc:14928
Change-Id: If2c848489bf94a1f7a5ebf90d2886d90c202c7c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/336240
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41687}
ReadStringView() is a simple alternative to ReadString() but doesn't
involve a heap allocation for a new std::string.
Using the new methods in one place to start with.
Bug: none
Change-Id: I1100c6d258ffb4c8a31a46ba88a7f8bff9cf35cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/332120
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41533}
This completes the conversion of ByteBufferReader and ByteBufferWriter
to uint8_t.
No-Try: True
Bug: webrtc:15661
Change-Id: I4152a8a4fd2462282d4107b3c2eed19acc8b29b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/331640
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41403}
and make follow-on changes.
Bug: webrtc:15665
Change-Id: Ice646f88ba5a09d6a8d9ce70415d8a14d7050d3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/329781
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41393}
Also make it more convenient to use uint8_t array view for
interfacing to the class.
Bug: webrtc:15665
Change-Id: Ib671b5add79a48004133a6ecd99429534f7de1de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/328140
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41212}
Goal is to initialize peerconnections in Chromium using this based
field trial config until a proper config that doesn't rely on the
global field trial string can be used (https://crrev.com/c/4936314).
Change-Id: I3d006e2445ccc4880b73b564c8ad4408242d3696
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/323621
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#40941}
This patch adds a StunDictionary.
The dictionary has a reader and a writer.
A writer can update a reader by creating a delta.
The delta is applied by the reader, and the ack is applied by the
writer.
Using this mechanism, two ice agents can (in the future) communicate
properties w/o manually needing to add new code.
The delta and delta-ack attributes has been allocated at IANA.
Bug: webrtc:15392
Change-Id: Icdbaf157004258b26ffa0c1f922e083b1ed23899
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/314901
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40513}
This is to allow external tests to depend on it.
Bug: none
Change-Id: Ic8e2f864041d959f673e7f2c18eb563a13274dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298745
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39646}
Under the combined network/worker thread project, tasks
are unnecessarily posted to the same thread.
This CL reaps 90% overhead savings in sent packet feedback
as measured with Perfetto under a 49p Meet call.
The identity of the posted calls was uncovered with WebRTC/Chrome's
new location-aware tracing.
TESTED=presubmit + local Meet calls.
Bug: chromium:1373439
Change-Id: I0c43aa4de884831838747d52b21c45fd360106e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295780
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39484}
Keeping the headers to allow compatibility with current users
that expect the headers to be in that target before they are
also updated.
Bug: webrtc:9838
Change-Id: I8b1e88850958e92c043686587a37791f01860220
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290569
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39031}
This new class implements the existing FieldTrialsView interface,
extending it with the verification functionality. For now, the
verification will only be performed if the rtc_strict_field_trials GN
arg is set.
Most classes extending FieldTrialsView today have been converted to
extend from FieldTrialsRegistry instead to automatically perform
verification.
Bug: webrtc:14154
Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38453}
Also call out the places where it happens explicitly - these are places
that need to be redesigned.
Bug: chromium:1177125
Change-Id: I3237d028dbb22380e8fbf7cedb03e965d1fcf2aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279022
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38384}
