and those will be fixed after I fixed downstream.
Bug: webrtc:10335
Change-Id: Ie824b422b4240fbcdb5d7ee40ae9be91655abae7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/256111
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36274}
This is part of a large-scale effort to increase adoption of
absl::string_view across the WebRTC code base.
This CL converts the majority of "const std::string&"s in function
parameters under rtc_base/ to absl::string_view.
Bug: webrtc:13579
Change-Id: I2b1e3776aa42326aa405f76bb324a2d233b21dca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254081
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Xavier Lepaul <xalep@webrtc.org>
Reviewed-by: Anders Lilienthal <andersc@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36239}
This patch just refactors creation of P2P transport channel,
pushing down the IceTransportInit object rather than decomposing
it going down.
The IceTransportInit object will in subsequent patches be
extended with a field trial container.
Reason for splitting patch into this and subsequent is
to allow changes to internal factories.
Bug: webrtc:10335
Change-Id: Icc8b6e4142744b64d134bcb2d4a56777745db62b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/255602
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@{#36215}
This cl/
1) move WebRtcKeyValueConfig from api/transport to api/ directory.
2) add a test/ScopedKeyValueConfig (compare ScopedFieldTrials).
3) removes usage of webrtc::field_trial:: from the pc/ directory.
4) removes a few unused includes of system_wrappers/field_trial.h.
Bug: webrtc:10335
Change-Id: If29c07900dbe791050b0a5ad05332bedfad035f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253903
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36160}
This cl/ extends the RTCIceCandidateStats object with
network_adapter_type and vpn, so that it maps the underlying
WebRTC objects completly.
Bug: webrtc:13773
Change-Id: I5cf79972c60ca6bf2a127dc96fa90811263ba6fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253241
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36110}
This cl/ changes so that the RTCTransportStats bytes/packets
sent/recevied is computed in P2PTransportChannel. Previously
they were computed by aggregating over the Connections, but that
does not work when Connections are created and destroyed.
Bug: webrtc:13769
Change-Id: Ia97dfae70b5aced897d4813ec007ba61bc032f87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/253100
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36103}
This field trial sets a non-zero receive buffer on the media UDP socket
with the intention to result in less packet loss in situations when the
application can't read packets fast enough from the socket. This can be
due to e.g. external factors, e.g. operating system not scheduling the
application for a longer time, or due to internal factors, e.g. slow
processing, a long running garbage collector, and more.
The size as set as the field trial parameter, as e.g.
WebRTC-SetSocketReceiveBuffer/Enabled-250/ to set it to 250kb.
Bug: webrtc:13753
Change-Id: Iae38d0db0c595d6e0148a2fdeb85ee8895e90560
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252581
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Commit-Queue: Victor Boivie <boivie@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36080}
This reverts commit 44156fa024cbf12f052a35571ac91bc9907be6c3.
Reason for revert: Needed in order to revert https://webrtc-review.googlesource.com/c/src/+/249941, which introduced a crash
Original change's description:
> Remove workaround in AutoSocketServerThread that isn't needed anymore.
>
> Cleanup steps for the Connection class have changed as of:
> https://webrtc-review.googlesource.com/c/src/+/249941
>
> However, it turns out that the PortTest suite still needs it, so the
> workaround has migrated to there.
>
> Bug: none
> Change-Id: Ia68f47b6c65b3a8fd5e8c04d70a43d15ba1a6422
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250223
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35894}
Bug: none
Change-Id: I13a4a79ebcb864054d14c1ba7726e18e044e3bd4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/252542
Auto-Submit: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36076}
in the case of an ip address the hostname() call will return
that. This may also avoid leaking IP addresses from DNS resolutions
and is more similar to the url originally passed into the
peerconnection (but will for example produce a fully formed url and
resolves the port if none was given).
BUG=webrtc:13652
Change-Id: I000c66f7988b4b205e38c4dde5b888e48d8f6a0f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250202
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35898}
Cleanup steps for the Connection class have changed as of:
https://webrtc-review.googlesource.com/c/src/+/249941
However, it turns out that the PortTest suite still needs it, so the
workaround has migrated to there.
Bug: none
Change-Id: Ia68f47b6c65b3a8fd5e8c04d70a43d15ba1a6422
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250223
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35894}
STUN servers don't do allocate requests, just binding requests.
Fix the description of onicecandidateerror accordingly.
BUG=None
Change-Id: I5698f23b50de46eb76175d1af5e88b605cd152f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/250142
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com>
Cr-Commit-Position: refs/heads/main@{#35876}
Following [1], add many more checks for safe access to member variables.
This change is effectively a no-op, but landed separately from the
earlier change that's smaller but contains a fundamental assumption
gleaned from the implementation (and its use).
[1]: https://webrtc-review.googlesource.com/c/src/+/249942
Bug: webrtc:11988
Change-Id: I1568e2160c9faa6993c5b68044312f83d00e4815
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249943
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35850}
Add a field trial "WebRTC-DscpFieldTrial"
that allows user to set any int value to be
used as tagging. This tag value will be used
for all packets on the PeerConnection,
whether they are audio, video, data or ICE
e.g WebRTC-DscpFieldTrial/override_dscp:40/
see https://webrtc.googlesource.com/src/+/b477fc73cfd2f4c09bb9c416b170ba4b566cecaf/rtc_base/dscp.h
for names of popular ints.
Bug: webrtc:13622
Change-Id: Iedbedd0f918100259678eb5bc083c9bf89b343b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249786
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35848}
Make sure that instances are always created+deleted on the
network thread.
Bug: webrtc:11988
Change-Id: I4fb5dd5bd14768d89ca78b348988a797fcdd130a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249942
Reviewed-by: Niels Moller <nisse@webrtc.org>
Auto-Submit: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35842}
https://crrev.com/b477fc73cfd2f4c09bb9c416b170ba4b566cecaf added a
cost for VPN, but I forgot to fix this method at the same time.
The VPN cost has luckily(?) not yet been rolled out, so no harm done!
Bug: webrtc:13097
Change-Id: I1e513eb0c1d5ca7a8efd184b5cf6ceeca7112cca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249603
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35827}
This is a safe cleanup change since top-level const applied to
parameters in function declarations (that are not also
definitions) are ignored by the compiler. Hence, such changes do
not change the type of the declared functions and are simply
no-ops.
Bug: webrtc:13610
Change-Id: Ibafb92c45119a6d8bdb6f9109aa8dad6385163a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249086
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35802}
On some networks, it's possible to have a DNS search domain pushed that
might make what is an invalid hostname succeed a DNS query.
In this case, invalid.com has a wildcard DNS entry and it would make this
test fail. Using a FQDN instead prevents search domains from being used.
Bug: none
Change-Id: I013f012db147b9c428b18d60e94a615153f199a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237810
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35355}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
Follow up to https://webrtc-review.googlesource.com/c/src/+/236260,
after removing use of deprecated methods/fields downstream.
Bug: webrtc:12132
Change-Id: Ic954c5c6785f30e327353e609fd5d55396f15810
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35305}
The AsyncListenSocket::SetOption method then gets unused, and can be
deleted.
Bug: webrtc:13065
Change-Id: Idcf70a75b96036290fdceff6e0f96a8d5617f87f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236580
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35302}
Removes the ability to accept nonencrypted answers to encrypted offers.
Fixes some logic around bundled sessions and requirement for
transport parameters.
Bug: webrtc:11066
Change-Id: I56d8628d223614918a1e5260fdb8a117c8c02dbd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236344
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35298}
This reverts commit 3b18208f13e85b356e61a95c0a261e9781403743
and is the third attempt at removing stun origin support
Bug: webrtc:12132
Change-Id: Ic41a6d011fb6239907a257cc4c81ec4d2923dc4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236260
Reviewed-by: Taylor Brandstetter <deadbeef@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Christoffer Jansson <jansson@google.com>
Cr-Commit-Position: refs/heads/main@{#35294}
This is a reland of b141c162ee2ef88a7498ba8cb8bc852287f93ad2
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
Bug: webrtc:13065
Change-Id: I88bebdd80ebe6bcf6ac635023924d79fbfb76813
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35260}
Only affects turn server. Refactored to wrap sockets with SSLAdapter
after Accept, using the SSLAdapterFactory to hold needed configuration.
Bug: webrtc:13065
Change-Id: I5df65aad5728d8d40d95b22db6398a573ec7a36f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235823
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35258}
This reverts commit b141c162ee2ef88a7498ba8cb8bc852287f93ad2.
Reason for revert: Breaking WebRTC rolls. See https://ci.chromium.org/ui/b/8832847811929676465 for an example failed build.
Original change's description:
> Take out listen support from AsyncPacketSocket
>
> Moved to new interface class AsyncListenSocket.
>
> Bug: webrtc:13065
> Change-Id: Ib96ce154ba19979360ecd8144981d947ff5b8b18
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/232607
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35234}
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:13065
Change-Id: Id5d5b35cb21704ca4e3006caf1636906df062609
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235824
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35249}
This is a reland of ba29ce320fe1f9ac69b0ff8eb50fbe402c2912a6
readding the origin to the CreateRelayPortArgs structure to not break
downstream tests yet:
https://webrtc-review.googlesource.com/c/src/+/235300/1..2
Original change's description:
> remove stun origin support
>
> Bug: webrtc:12132
> Change-Id: I0f32e6af77e0c553b0c3b0d047ff03e14c492b31
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234384
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#35202}
Bug: webrtc:12132
Change-Id: Ied840b59bb7c9497e98f9b80eb0a54d30008a40f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235300
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35220}
