Using simulated time implementation of TaskQueue with TaskQueueForTest.
Bug: webrtc:10365
Change-Id: I9f853133aa9fcd5858a5e2e1e0f47ab1a6a5ca18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127292
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27081}
Implicit bool conversions behave wierdly in a bunch of cases, so let's make it explicit.
Bug: None
Change-Id: I15933e90d57c57218eed9608407aace5a640a6ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/127284
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27076}
That will avoid linkage error of targets that depend
on global_task_queue_factory without depending on rtc_task_queue
Bug: webrtc:10191
Change-Id: Ia67ea0a3abb715e28160e4e376133cc0309b14e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/126621
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27053}
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}
Ignore rtc_link_task_queue_impl flag,
instead use build_with_chromium for custom chromium implementation injection
This changes TaskQueue implementation used in webrtc fuzzers in chromium:
from own webrtc implementation to chromium's.
Bug: webrtc:10191
Change-Id: I63be28b680ae8ea8ee1dbf0c699263c392ce29d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/125196
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26977}
It seems native mutex performance has improved considerably on Mac
lately, primarily by switching to a different default scheduling
policy. For safety, set this policy explicitly.
The special implementation previously used on Mac is still faster but
suffers a problem when used on realtime audio threads, where they will
not get rescheduled as quickly as when using native mutexes.
Bug: webrtc:10373
Change-Id: Iabf97afc5c2609096331bba0199f433fd26b68b2
Reviewed-on: https://webrtc-review.googlesource.com/c/125186
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26948}
The header "rtc_base/fake_mdns_responder.h" was not self contained,
and it was relying on order includes to get all the types it needs.
This CL makes it self-contained and removes an unneeded #include.
Bug: None
Change-Id: Ie6c584a169ef884d79e436e51c2e72236b0d4c7a
Reviewed-on: https://webrtc-review.googlesource.com/c/125184
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26938}
This is a partial cleanup there is more work to do here. Essentially I am just
moving things from static to anonymous namespaces and reordering things to
make more sense. I have removed some old microsoft compiler warning
supressions which I believe are not required anymore.
After this BIO should be refactored to use proper style.
Bug: webrtc:9860
Change-Id: I8419be002d8f412dd89f37f3b865794792ccf559
Reviewed-on: https://webrtc-review.googlesource.com/c/120863
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26912}
This method allows asynchronously posting tasks, in the form of
functors to be invoked, on the thread represented by rtc::Thread.
This CL removes PostMessageWithFunctor(), putting the implementation of
it directly into rtc::Thread::PostTask(), and moves & updates the test
coverage to thread_unittest.cc.
Bug: webrtc:10294, webrtc:10293
Change-Id: Ic6cc3e2533a4f7aaff141aff28e9bb3908ee3c96
Reviewed-on: https://webrtc-review.googlesource.com/c/124701
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26894}
This change simply adds const to all the variables that can use it. It seemed
like a good idea to enforce const correctness where possible in the TLS stack.
Bug: webrtc:9860
Change-Id: Iabfe1e26647b0c21e2f209e0e0f96d0ec7465e7a
Reviewed-on: https://webrtc-review.googlesource.com/c/124623
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26865}
There seems to be a race caused by the libevent wrapping TaskQueue
implementation when reposting a repeated task at destruction time. This
race results in the posted task being leaked according to asan.
Bug: webrtc:10278
Change-Id: Ida40b884547f3f789a804ca0ab3ce36982a4d68e
Reviewed-on: https://webrtc-review.googlesource.com/c/121424
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26839}
It was previously possible to escape the sandbox by calling
rtc::Thread::SetAllowBlockingCalls(true).
This CL only removes the loophole on non-Android builds, because we
still have old Android code that relies on it. We expect that code to
go away soon-ish, though.
Bug: webrtc:9987
Change-Id: Ida96400d0abe430af4c2046284795d37d64f6613
Reviewed-on: https://webrtc-review.googlesource.com/c/123523
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26792}
(reverted in https://webrtc-review.googlesource.com/c/src/+/123182/1)
Original cl description:
Always offer transport sequence number header extension for audio
If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.
Patchset 3 contain the only change:
Add the field trial WebRTC-Audio-SendSideBwe to call/rampup_tests.cc
TBR: srte@webrtc.org,ossu@webrtc.org
Bug: webrtc:10309 webrtc:10286
Change-Id: I2c1224e8a9fab52c1030369c1364686322e88a0f
Reviewed-on: https://webrtc-review.googlesource.com/c/123183
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26706}
This function is used to post messages onto rtc::Threads. The thread
invokes the functor without blocking the calling thread. Messages posted
in this way are executed in the order that they were posted. This is
meant to work as the equivalent of "thread->PostTask()" in Chromium.
Note: AsyncInvoker currently does something similar but it is more
cumbersome to use (somebody has to create it and own it and make sure
not to destroy it while tasks are pending or else they're cancelled). It
also comes with a fundamental flaw: You cannot destroy the AsyncInvoker
from within the functor (this results in a neverending Wait). This makes
the AsyncInvoker not suitable for implementing "destructor traits"
amongst other things.
This CL will allow us to easily add "PostTask()" to rtc::Thread or add
support for DestructorTraits, which is especially useful when you have a
reference counted object that is referenced from multiple threads but
owns resources that has to be destroyed on a particular thread.
Blocking invokes are forbidden in Chromium but WebRTC performs them
frequently. Being able to perform the equivalent of PostTask() is a
good thing.
Bug: webrtc:10293
Change-Id: Ie2a612059a783f18ddf98cff6edb7fce447fb5be
Reviewed-on: https://webrtc-review.googlesource.com/c/121408
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26704}
This reverts commit fd965c008c7bc395bb276f260262ac11ccd25406.
Reason for revert: Cause test failure.
Original change's description:
> Always offer transport sequence number header extension for audio
>
> If the extension is negotiated, it will only be used if
> the field trial WebRTC-Audio-SendSideBwe is enabled.
> This allows simpler experimentation if it should be used or not.
>
> Bug: webrtc:10309 webrtc:10286
> Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
> Reviewed-on: https://webrtc-review.googlesource.com/c/122542
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#26689}
TBR=ossu@webrtc.org,sprang@webrtc.org,srte@webrtc.org,perkj@webrtc.org
Change-Id: I1b7d3fa5c282a5bf049ca54695ad16c8278a2698
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10309 webrtc:10286
Reviewed-on: https://webrtc-review.googlesource.com/c/123182
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26703}
If the extension is negotiated, it will only be used if
the field trial WebRTC-Audio-SendSideBwe is enabled.
This allows simpler experimentation if it should be used or not.
Bug: webrtc:10309 webrtc:10286
Change-Id: I797e6f14c06d46189e40f6d09805c2e09afc015b
Reviewed-on: https://webrtc-review.googlesource.com/c/122542
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26689}
This quality boost means that we sometimes drop a _lot_ of frames in the
base layer. It also interacts poorly with the bitrate adjuster since
even if frames are dropped they are often over-sized.
The setting still leaves the current behavior as default, but can be
changed using the WebRTC-VideoRateControl field trial.
Bug: webrtc:10155
Change-Id: I1a92ec69bab61b5148fe9d8bc391ac5ee1019367
Reviewed-on: https://webrtc-review.googlesource.com/c/122840
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26659}
This replaces the functionality provided by
AudioPriorityBitrateAllocationStrategy, removing the need provide that
component via injection in all clients using audio bitrate priority.
Bug: webrtc:10286
Change-Id: I3bafab56d24459d9d27dc07abffdc8538440a346
Reviewed-on: https://webrtc-review.googlesource.com/c/121402
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26651}
