This patch adds an interface_id property
to rtc::Network. It is an enumeration of the
interface names that are present.
This enables a local ICE agent to keep track
of which connections are using which interfaces,
something that is useful for predicting how
connections behave.
This is part 1 of https://webrtc-review.googlesource.com/c/src/+/85520
BUG: webrtc:9446
Change-Id: Ia6ec1f14ac240799fb1be49d67d82e2733e87acf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171061
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30882}
This patch extends the NetworkRoute struct with more information
about local/remote endpoints. It adds
- adapter type
- adapter id
- relay
(previously it was "only" network_id)
The patch leaves the {local/remote}_network_id fields
around and populated since downstream projects depend
on them. They will be removed once they have migrated.
OWNER: srte@ call/ test/
OWNER: asapersson@ video/
OWNER: hta@ p2p/ pc/ rtc_base/
BUG: webrtc:11434
Change-Id: I9bcec385b40d707db385fef40b2c7a315dd35dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170628
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30848}
This reverts commit af1f8655b2cb69af382396ea642eb0a2bf04bb4d
Landing the change with default set to
"enabled" (DTLS 1.0 will continue to work by default),
so that flipping the default can be a separate CL.
Original change's description:
> Revert "Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC."
>
> This reverts commit 7276b974b78ea4f409d8738b1b6f1515f7a8968e.
>
> Reason for revert: Changing to a later Chrome release.
>
> Original change's description:
> > Disable DTLS 1.0, TLS 1.0 and TLS 1.1 downgrade in WebRTC.
> >
> > This change disables DTLS 1.0, TLS 1.0 and TLS 1.1 in WebRTC by default. This
> > is part of a larger effort at Google to remove old TLS protocols:
> > https://security.googleblog.com/2018/10/modernizing-transport-security.html
> >
> > For the M74 timeline I have added a disabled by default field trial
> > WebRTC-LegacyTlsProtocols which can be enabled to support these cipher suites
> > as consumers move away from these legacy cipher protocols but it will be off
> > in Chrome.
> >
> > This is compliant with the webrtc-security-arch specification which states:
> >
> > All Implementations MUST implement DTLS 1.2 with the
> > TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256 cipher suite and the P-256
> > curve [FIPS186]. Earlier drafts of this specification required DTLS
> > 1.0 with the cipher suite TLS_ECDHE_ECDSA_WITH_AES_128_CBC_SHA, and
> > at the time of this writing some implementations do not support DTLS
> > 1.2; endpoints which support only DTLS 1.2 might encounter
> > interoperability issues. The DTLS-SRTP protection profile
> > SRTP_AES128_CM_HMAC_SHA1_80 MUST be supported for SRTP.
> > Implementations MUST favor cipher suites which support (Perfect
> > Forward Secrecy) PFS over non-PFS cipher suites and SHOULD favor AEAD
> > over non-AEAD cipher suites.
> >
> > Bug: webrtc:10261
> > Change-Id: I847c567592911cc437f095376ad67585b4355fc0
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125141
> > Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> > Reviewed-by: David Benjamin <davidben@webrtc.org>
> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#27006}
>
> TBR=steveanton@webrtc.org,davidben@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: webrtc:10261
> Change-Id: I34727e65c069e1fb2ad71838828ad0a22b5fe811
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/130367
> Commit-Queue: Benjamin Wright <benwright@webrtc.org>
> Reviewed-by: Benjamin Wright <benwright@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27403}
Bug: webrtc:10261
Change-Id: I28c6819d37665976e396df280b4abf48fb91d533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169851
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30733}
Adding :: before rtc allow us to use the macro in nested rtc namespace for external components like
namespace xxxxxxx {
namespace rtc {
RTC_CHECK(true);
}
}
Bug: webrtc:11400
Change-Id: I79349b847c3fce8197c82aec31b672a1a16e5388
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169683
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30684}
Selling point is that it never touches the heap. Intended use case is
cheaply returning a variable, bounded, and small number of things from
a function.
Specifically, there are situations where we'd like to return things like
ArrayView<ArrayView<float>>
where we currently have to allocate an array of ArrayView<float> for
the outer ArrayView to point to, which is a bother; however, although
the outer ArrayView is of variable size, that size is statically
guaranteed to not exceed some small constant. After this CL, we'll be
able to instead return
BoundedInlineVector<ArrayView<float>, kSmallConstant>
which is much more convenient. We already had the option of returning e.g.
std::vector<ArrayView<float>>
but that would bloat our binary with code to handle heap allocations
in places we'd rather be lean and mean.
https://godbolt.org/z/r-vcPj demonstrates that the overhead compared to
a raw C array + a size is ~zero.
Bug: webrtc:11391
Change-Id: Ifb6d937193052588be641aa62cc67ba0ec64ded6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168944
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30663}
This avoid duplication. As part of this moving the overhead calculation
to the IP address class so it's easier to find and more natural to use.
Bug: webrtc:9883
Change-Id: If4d865f445bc1a302572896932966ce30294e339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169445
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30657}
Move definition of AlignedArray to the only code using it, the
test-only LappedTransform class, and delete unused methods.
Bug: webrtc:6424, webrtc:9577
Change-Id: I1bb5f57400f7217345b7ec7376235ad4c4bae858
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168701
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30576}
With current congestion window pushback, when congestion window is filling up, it will reduce bitrate directly and encoder may reduce encode quality, resolution, or framerate to adapt to the allocated bitrate, the behavior is depending on the degradation preference.
This change enable congestion window to only drop frames to reduce bitrate (when needed) instead of reduce general bitrate allocation.
Bug: webrtc:11334
Change-Id: I9cf5c20a0858c4d07d006942abe72aa5e1f7cb38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168059
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30483}
This CL only includes the necessary changes in PhysicalSocketServer,
and doesn't include the Java or Objective C API.
Note that this is doing exactly the same thing as UDPSocketPosix
in chromium.
BUG=webrtc:5658
Change-Id: I295455eaccba2a83cdd1bc55848f325c310f8d32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168260
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30478}
The AsyncTCPSocket is an AsyncPacketSocket which means it
emulates UDP-like (packet) semantics via a TCP stream. When
sending, if the entire packet could not be written then the
packet socket should indicate it wrote the whole thing and
flush out the remaining later when the socket is available.
The WriteEvent signal was already wired up but was not getting
fired (at least with the virtual sockets) since it would not
call Send() enough on the underlying socket to get an
EWOULDBLOCK that would register the async event.
This changes AsyncTCPSocket to repeatedly call Send() on the
underlying socket until the entire packet has been written
or EWOULDBLOCK was returned.
Bug: webrtc:6655
Change-Id: I41e81e0c106c9b3e712a8a0f792d28745d93f2d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168083
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30449}
Since RTC_DCHECK was made constexpr compatible, we can now
make the unit classes fully constexpr.
Bug: webrtc:9883
Change-Id: I18973c2f318449869cf0bd45699c41be53fba806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167722
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30403}
This makes it easier to maintain consistency between real time
and simulated time modes.
The RealTimeController is updated to use an explicit main thread,
this ensures that pending destruction tasks are run as the network
emulator goes out of scope.
Bug: webrtc:11255
Change-Id: Ie73ab778c78a68d7c58c0f857f14a8d8ac027c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166164
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30342}
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.
Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.
CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn
Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).
[1] - https://gn-review.googlesource.com/c/gn/+/6860
Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
This avoids integer overflows when size_t is 32 bits, and conforms
to style guide recommendations to avoid unsigned integers.
Also add tests for overflow on RateStatistics accumulator.
Bug: webrtc:11247
Change-Id: Ifa0db567f41bbcf3ec46d89ab888f2ed9d03f3f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163991
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30268}
The test sometimes failed because thread creation on some platforms on
internal tests may take too much work.
Now checks are less strict.
Bug: none
Change-Id: Ibd3df02bda26b0c5e804360a909c61afa760b3bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165960
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30267}
This prepares for introducing a simulated time rtc::ThreadManager
implementation that will run on a single underlying thread.
Bug: webrtc:11255
Change-Id: I793128cc0b8e649a3675914de67dfee3298b446a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165765
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30256}
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format
After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.
This primary benefit of this change is a small reduction in binary size.
Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
Since rtc::Thread is the only class inheriting from rtc::MessageQueue
and most members of MessageQueue are public or protected the split is
not adding much value. In preparation for future cleanup, this cl merges
the two classes.
Bug: webrtc:9883
Change-Id: Ia0efb4349f66f653aa34fa4d244998f187e3ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30235}
The test was failing (in a flaky fashion, interestingly), comparing:
* 50.5 whose mantissa is:
1001010000000000000000000000000000000000000000000000
* with 50.500000000000036 whose mantissa is:
1001010000000000000000000000000000000000000000000101
since EXPECT_DOUBLE_EQ() only allows 4 ULPs difference.
We don't need this kind of precision.
Bug: webrtc:11134
Change-Id: I811178b0762dbcd61d4f2d3f047ea0b59847fa57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165761
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30230}
Add missing #include to fix some build error on `winuwp` with some code
using rtc::dchecked_cast<> under an `#ifdef (WINUWP)`, resulting in an
undefined symbol error.
Bug: webrtc:11194
Change-Id: Iad9e74c3e92ed6cf1461f34cdd9329d13f5d62e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161721
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30224}
