This cl/ fixes a race condition with the recent additions
to NetworkMonitorAutoDetect (getAllNetworksFromCache).
The getAllNetworksFromCache-feature uses the by the Android team preferred way of
enumerating networks, i.e to register network listeners.
This however introduces a unpleasant race condition like this:
1) network.cc discover rmnet0
2) BasicPortAllocator tries to create UDP port on rmnet0
- This fails as BindSocketToNetwork requires a android handle.
3) NetworkMonitorAutoDetect gets callback with rmnet0
4) BasicPortAllocator tries to create TCP port on rmnet0
- This succeeds.
5) Since rmnet0 has one working port, there will not be
any new ports created on that network
=> We end up with a TCP only connection :(
---
By impl. IsAdapterAvailable, we make sure that the network
will not be used by BasicPortAllocator (or anyone else!)
until we support binding to it.
The IsAdapterAvailable was implemented for IOS,
and has test coverage using FakeNetworkManager.
This cl/ is default enabled with the kill-switch
WebRTC-AndroidNetworkMonitor-IsAdapterAvailable.
Bug: webrtc:13741
Change-Id: I7c2cb7789660fd2e082c214d00e50c894666b07c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257400
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36406}
When resetting several streams in sequence, only the first stream will
be included in the first RE_CONFIG chunk as it's created eagerly
whenever someone calls ResetStreams. The remaining ones are queued as
pending. When the first request finishes, the remaining ones should
continue to be processed, but this wasn't done prior to this commit.
This would only happen if two streams would be reset with shorter time
between them than a RTT, so that there would be an outstanding request
forcing the second reset to be enqueued.
Bug: chromium:1312009
Change-Id: Id74b375d1d1720406a3bca4ec60df5780ca7edba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257306
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36404}
Change adapts the `base_capturer_pipewire` so that a portal can be
injected in the capturer. This allows the remoting to inject its
own portal for the purpose of capturing desktop stream as long
as the injected portal provides implementation of the new interface
that is added as part of this change.
Additionally, a method has been exposed on the capturer to get
details about the portal session so that the remoting
implementation can use the same underlying session for controlling
inputs on the remote host.
Finally, desktop capturer interface is extended with a generic
method `GetMetadata` that is used to retrieve session related
information by CRD and relay it over to its input injector. Clients
provide override for the method and it eventually invokes the
underlying `GetSessionDetails` method on the portal instance.
Bug: chromium:1291247
Change-Id: I0dbd154eb16d4149f967c4a818eea51e7e6eb9a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257000
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Salman Malik <salmanmalik@google.com>
Cr-Commit-Position: refs/heads/main@{#36399}
Default holdback-window for non-prio packets is now 5ms, or the expected
pacing time for 3 packets if lower.
This brings wakeup frequency in line with legacy pacer at medium to low
packet rates.
Bug: webrtc:10809
Change-Id: I4045c40ae6b6d50f1ea049f3a26768023f77ec3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257301
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36397}
This was available in the beginning, as a way to increase the chance of
a message sent with partial reliability to be delivered, by avoiding it
to be fragmented in too small fragments.
This however added a few downsides:
* Packet efficiency goes down, as the entire MTU isn't always used
* Complexity increases when adding message interleaving, since if one
stream refuses to produce messages, but there is another stream with
a very small message that could fit in its place, it should be used
instead.
Removing this feature altogether is much easier. It's hard to defend.
Bug: webrtc:5696
Change-Id: Ie2f296e052f4a32a281497d379c0d528a2df3308
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257168
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36396}
RFC8831, section 6.7 states that closing a data channel MUST be signaled
as resetting an outgoing stream, and that will ensure that all messages
are either delivered or abandoned before the stream is reset. In the
current implementation, dcSCTP has opted to abandoned any queued
messages that haven't been partially sent.
And this CL simply adds more documentation around this choice. It's
subject to change and a client implementation shouldn't depend on any
such behavior as the RFC allows the implementation to decide.
Bug: None
Change-Id: I60305fe396a6a3f494d823c71e092acfeb6075b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257167
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36395}
This is to aid with catching issues whereby a connection object might
have a bad reference back to a port object, e.g. inside of an async
callback.
Bug: webrtc:13892
Change-Id: I56503fedc2865919713b10f236ce023554c68ded
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257164
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36394}
This use case was missing test coverage and sufficient comments in the
code.
Bug: None
Change-Id: I95b54a64f714b68a347fdbeef79eb38e715adbc3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257166
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36393}
This cl adds a new field trial parameter to WebRTC-Bwe-EstimateBoundedIncrease that allows the delay based estimate to immediately increase to the upper link capacity estimate.
Bug: none
Change-Id: I875121a41f65cc8e76bb87bbf421731ba6b4551b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257142
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36392}
* tools_webrtc/PRESUBMIT.py is only checking the licence which is already done here:
38f35db4d4:PRESUBMIT.py;l=913;bpv=1;bpt=0;drc=4fc9bd9f69a0d88889d86d0cc9f8e27406e8a342
* sdk/android/PRESUBMIT.py was added before 'git cl format' was required from the root PRESUBMIT.py:
https://codereview.webrtc.org/2377113003
Bug: webrtc:13895
Change-Id: Ia5ea2529c36ceebfd7d4e6a6a72352bd30c573b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257280
Reviewed-by: Christoffer Jansson <jansson@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#36391}
This is used until the first RTT measurement becomes available.
100ms is a reasonable default and used in other places.
Bug: webrtc:10178
Change-Id: I14f530504a4866fbe75f025dfe184fd6e296b75e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256861
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36384}
`enable_non_sender_rtt_` is updated a few lines below.
This seems to have been missed in the code review.
Bug: webrtc:12951, webrtc:13853
Change-Id: Idc06421362f6b2d831b5a828f296142aab9a46e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256860
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36380}
This CL changes the default speed settings for TL0/TL[1-2] from
7/8 to 9/9 at 1080p resolutions and up. We also disable the denoiser
at these resolutions.
Settings can be overriden using existing WebRTC-VP9-PerformanceFlags
field trial.
Bug: webrtc:13888
Change-Id: I70f19efdace88d70bbb90bc6dd5149653eb079c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257141
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36379}
Create FieldTrials class for setting the field trials from a string
in a program. We can later e.g add a builder class where one
can add key/value pairs.
This class is supposed to replace
webrtc::field_trial::InitFieldTrialsFromString.
No-Try: True
Bug: webrtc:10335
Change-Id: I17f45e401102fddda50ca7c4a04bea2f1cb87788
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256973
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36376}
Ironic :( The "field trial guy" constructing a invalid string,
if only there would have been a builder instead...
I tested the code several times...but not with debug build...
Bug: webrtc:13741
Change-Id: If3caad0f5533fc150ffd6a34a89ab84f3f0264aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256979
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36370}
The Frequency implementation does not allow for nominators as large as
those that can occur in consecutive RTP timestamps, so use double math
instead.
Bug: chromium:1310611
Change-Id: I3b239e1b84043470ca29da06728b42cd4552300f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256978
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36368}
Needed to migrate downstream code that needs to call new signature of
BasicNetworkManager::GetNetworks(). And similarly for
GetAnyAddressNetworks.
Bug: webrtc:13869
Change-Id: I8b9e842e74e290662c0713846dc29ac739d76ba2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256977
Auto-Submit: Niels Moller <nisse@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36366}
This explores the theory that targets that have no files, just
dependencies, are unnecessary.
Bug: webrtc:13805
Change-Id: I1feb50cf3886128031af8970eae361e35fb052c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256974
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36363}
convert rtc_base/network and collateral.
This also remove last usage of system_wrappers/field_trials
in p2p/...Yay!
Bug: webrtc:10335
Change-Id: Ie8507b1f52bf7f3067e9b4bf8c81a825e4644fda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256640
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36357}
