This was caused by change 231229 and was later caught when reviewing the
code. The rtt variable was accidentally re-used for another purpose, and
then assumed to still be used to represent the rtt.
There have been no issues found with this re-use, but it was wrong.
Bug: webrtc:12614
Change-Id: If1a180315cf833e293f78c54c3c3b29394a19a20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232610
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35064}
In callers where it's non-trivial to explicitly pass the right
SocketFactory, pull the call to rtc::Thread::socketserver() into the
caller, with a TODO comment.
Bug: webrtc:13145
Change-Id: I029d3adca385d822180e089f016c3778e0d4fd0c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231227
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35063}
Prior to this commit, the SCTP association could terminate due to too
many retransmission attempts when there is a long duration of packet
loss. The RTCPeerConnection wouldn't terminate, and when the network
later recovers (possibly using a different ICE candidate), it would be a
RTCPeerConnection with media, but without DataChannels.
This commit will make the dcSCTP library never abort by itself when
there are too many retransmissions. It will also put a cap on the retry
duration so that it will do a retry every three seconds (or lower).
Bug: webrtc:13129
Change-Id: I08162ea20d6a60aa0eae2717966d9a2ddba8fc22
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232540
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35061}
It's useful for other parts of WebRTC and there is no real reason why
it should be located in net/dcsctp.
Bug: None
Change-Id: Iccaed4e943e21ddaea8603182d693114b2da9f6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232606
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35055}
This one is frequently accessed - Mainly by ::CreateReportBlocks and
is visible in performance profiles (although not very much).
By using webrtc::flat_map, better data cache locality is expected.
Bug: webrtc:12689
Change-Id: Ic2ebcad806788074b2b4cb244a25395a48df1852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232541
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35054}
Small step towards using separate classes for TCP server sockets.
Added a new test-only class AsyncStunServerTCPSocket needed
for unit tests of AsyncStunTCPSocket.
Bug: webrtc:13065
Change-Id: I7d9713983d8f6b30aa3d3e7442bb34ea48b815eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232324
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35047}
Follow-up to https://webrtc-review.googlesource.com/c/src/+/232260:
there's a second link to the obsolete chromium-cpp.appspot.com in the
same file that was missed.
Also updates the link to be more precise: the new markdown file has
anchors on individual entries, not just the section headers.
R=danilchap@webrtc.org
Bug: chromium:1243839
Change-Id: I17918d155aacf3465a46fd674a598139a0870165
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232560
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35043}
Unlike ReadBits, ConsumeBits doesn't limit number of bits it may advance,
and thus should work when that number is close to the integer limit
Bug: chromium:1250730
Change-Id: Ia7847869ef9d3fc16450d572c9e2be6e1aa36741
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232332
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35042}
Ozone is default now in Chromium and non-Ozone/X11 (aka use_x11) is
deprecated. During the transition period use_x11 == ozone_platform_x11.
Bug: chromium:1096425
Change-Id: Ie3643360ec6607796533054bdedf8e2bb8e7e749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231650
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35040}
This allows build targets that need only HandoverReadinessStatus
to depend only on public:socket, and not on socket:dcsctp_socket.
Bug: webrtc:13154
Change-Id: I29f41910cdb5baed96b57fd7284b96fc50a56ba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232331
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35037}
dcSCTP library users can set their custom
g_handover_state_transformer_for_test that can serialize and
deserialize the state. All dcSCTP handover tests call
g_handover_state_transformer_for_test. If some part of the state is
serialized incorrectly or is forgotten, high chance that it will
fail the tests.
Bug: webrtc:13154
Change-Id: I251a099be04dda7611e9df868d36e3a76dc7d1e1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232325
Commit-Queue: Sergey Sukhanov <sergeysu@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35035}
This defaults the calculation landed in cl 196502. The less readable legacy calculation method will be deleted in a future CL.
Bug: none
Change-Id: Ida02a5208e354835b964c69355ad1e9d5bba18aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/231956
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35027}
Thanks to the elimination of `ExperimentalNs`, there is no need anymore
to pass `webrtc::Config` to build APM.
Hence, `AudioProcessingBuilder::Create(const webrtc::Config&)` is also
removed.
Bug: webrtc:5298
Change-Id: I0a3482376a7753434486fe564681f7b9f83939c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232128
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35025}
Follow up for https://webrtc-review.googlesource.com/c/src/+/232061/5. Updated mac M1 tests that was missed because they are not part of CQ
Bug: b/199885455
Change-Id: I77618ac2869ba601f322857f4391b63220d20252
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232220
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35021}
The new name more accurately reflects the intent of the actual implementation.
Bug: none
Change-Id: I3d2aeb561104165f9f9879854a4a210730e02ff5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232130
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Fanny Linderborg <linderborg@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35020}
This CL is for the same behavior as before [1], to emit the NSError
when an application is not using the RTCAudioSession lock correctly.
[1] https://webrtc-review.googlesource.com/c/src/+/207432
Bug: webrtc:13091
Change-Id: I031b0e963d33c92ce1af7a306edfa6be005e043d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229461
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Byoungchan Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#35018}
This CL ensures that each DesktopFrame's updated-region is expressed in
the frame's own coordinates, where the top-left is always (0, 0).
For example, DesktopFrame::GetFrameDataAtPos() and its callers use
this coordinate system.
Previously, whenever a RANDR monitor with a non-zero offset was
selected, ScreenCapturerX11 would hit some DCHECKs when trying to
copy pixels from previous frames, or when capturing new pixels into
them from XDAMAGE regions.
Bug: None
Change-Id: I7b2e8d0449359ee7b263ad60af193e2bf89aa1f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232085
Reviewed-by: Joe Downing <joedow@chromium.org>
Commit-Queue: Joe Downing <joedow@chromium.org>
Cr-Commit-Position: refs/heads/main@{#35017}
When there is no outstanding data, then next TSN to allocate should
always be one more than what the client has last ACKed.
Bug: None
Change-Id: Ieb8b5b23912d77d96fe3749fb53fd53652d97066
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232002
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35016}
