This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.
The mutex types supportable by webrtc::Mutex are
- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)
In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.
The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.
Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.
Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
This change removes dependency on SignalThread which is a
heavy user of re-entered mutexes, and is a step to being
able to delete SignalThread.
The new AsyncResolver is based on task queue instead.
Bug: webrtc:11567, webrtc:7723
Change-Id: Iab125ccbc0fb9d72af44341e58f89b3868c952cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175910
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31361}
This reverts commit d0d55515c4b88a07446bc66a5c183f50ee896282.
Reason for revert: Relanding with workaround.
Original change's description:
> Revert "Use Windows 10 thread naming API"
>
> This reverts commit e35004dffb42dd96b8cf37b33c9a3af4a5fd376c.
>
> Reason for revert: Reverting while downstream issue is resolved.
>
> Original change's description:
> > Use Windows 10 thread naming API
> >
> > While profiling video chat in Chrome I noticed that some of the webrtc
> > threads were not named. This change adds conditional use of the thread
> > naming APIs. These thread names work even if you attach a debugger after
> > the thread is named, and they show up in ETW traces, for easier
> > profiling.
> >
> > The sctp_create_thread_adapter threads are still not named but since
> > those are in C files they would require a C++-with-extern-C interface
> > to fix, so I'm leaving them for now.
> >
> > Bug: webrtc:10745
> > Change-Id: I68f6aa780e2417ce706764d69e5b64cc48aba333
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175280
> > Commit-Queue: Tommi <tommi@webrtc.org>
> > Reviewed-by: Tommi <tommi@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31285}
>
> TBR=kwiberg@webrtc.org,tommi@webrtc.org,brucedawson@chromium.org
>
> Change-Id: Icf877afbd82918ebe0c42a93b8a763cdab9a73ce
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:10745
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175347
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31289}
TBR=kwiberg@webrtc.org,tommi@webrtc.org,brucedawson@chromium.org
Bug: webrtc:10745
Change-Id: I51ee413fd8a0ff62f6b8b2a11f546b2a70168842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175349
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31292}
This reverts commit e35004dffb42dd96b8cf37b33c9a3af4a5fd376c.
Reason for revert: Reverting while downstream issue is resolved.
Original change's description:
> Use Windows 10 thread naming API
>
> While profiling video chat in Chrome I noticed that some of the webrtc
> threads were not named. This change adds conditional use of the thread
> naming APIs. These thread names work even if you attach a debugger after
> the thread is named, and they show up in ETW traces, for easier
> profiling.
>
> The sctp_create_thread_adapter threads are still not named but since
> those are in C files they would require a C++-with-extern-C interface
> to fix, so I'm leaving them for now.
>
> Bug: webrtc:10745
> Change-Id: I68f6aa780e2417ce706764d69e5b64cc48aba333
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175280
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31285}
TBR=kwiberg@webrtc.org,tommi@webrtc.org,brucedawson@chromium.org
Change-Id: Icf877afbd82918ebe0c42a93b8a763cdab9a73ce
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10745
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175347
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31289}
While profiling video chat in Chrome I noticed that some of the webrtc
threads were not named. This change adds conditional use of the thread
naming APIs. These thread names work even if you attach a debugger after
the thread is named, and they show up in ETW traces, for easier
profiling.
The sctp_create_thread_adapter threads are still not named but since
those are in C files they would require a C++-with-extern-C interface
to fix, so I'm leaving them for now.
Bug: webrtc:10745
Change-Id: I68f6aa780e2417ce706764d69e5b64cc48aba333
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175280
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31285}
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}
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}
`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}
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}
This change improves performance under high load by processing
all pending tasks each time the thread is woken up by libevent.
Additionally, the pipe used to wake up the TaskQueue thread now
not be written to if there's already a pending write on the pipe.
This fixes a bug where under high load the pipe write buffer can
fill and cause tasks to get dropped.
Bug: webrtc:11259, webrtc:8876
Change-Id: Ic82978c71bf9e9a25f281ca4775d46168d161d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165420
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30202}
Replace all usages of java_files with sources in gn files, and
automatically format.
This is in preparation for java_files being completely removed upstream
in favor of sources.
NOPRESUBMIT=true
Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
to generate less versions of the function template and FunctorMessageHandler helper
thus producing less binary size
Bug: None
Change-Id: Idbd6fb1e1f23b9b2dc4e4306a74ef11e74ba94cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161044
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29962}
This CL adds a dependecy on rtc_base/system:rtc_export to rtc_event but
only when built as part of Chromium (since rtc::Event should not be
used outside of WebRTC).
It also adds other missing RTC_EXPORTS.
Bug: webrtc:9419
Change-Id: Ib338004a5404a6b3c7929e146c29ad42572632cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159692
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29781}
This does not help the attached bugs, but it does allow greater control
over what JSON code is running where. Long-term, the JSON library used
for parsing configuration should likely be a library already present
in Chromium builds, to avoid duplication. And if that happens, then
WebRTC bug 9804 may be passé.
Note that this CL also sorts our poisons alphabetically.
Bug: chromium:895814, webrtc:9804
Change-Id: I70c3efe05a0eba9212895407f73978d8216df920
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158400
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29615}
Since rtc_base/ignore_wundef.h doesn't have any dependency, it is easy to
move it to its own target and allow its dependant to avoid to take a
dependency rtc_base:on rtc_base_approved.
Bug: webrtc:9419
Change-Id: I17f205b0cb2b21cad388b04e60082df9398dffdf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157428
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29548}
Static libraries don't guarantee that an exported symbol gets linked
into a shared library (and in order to support Chromium's component
build mode, WebRTC needs to be linked as a shared library).
Source sets always pass all the object files to the linker.
On the flip side, source_sets link more object files in release builds
and to avoid this, this CL introduces a the GN template "rtc_library" that
expands to static_library during release builds and to source_set during
component builds.
See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set
Bug: webrtc:9419
Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Nico Weber <thakis@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29525}
That allows to use SingleThreadedTaskQueueForTesting via TaskQueueBase interface
but still have access to test-only SendTask function.
Bug: webrtc:10933
Change-Id: I3cc397e55ea2f1ed9e5d885d6a2ccda412beb826
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156002
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29480}
rtc::CriticalSection has non-trivial destructor and thus
shouldn't be used for variable with static storage duration
Bug: None
Change-Id: I5b9d9036aa90eb0c652f6b17ea1162dea0362640
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156563
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29469}
Merge GlobalLock and GlobalLockPod, make member private.
annotate creation of all GlobalLocks with ABSL_CONST_INIT
Bug: None
Change-Id: I29abcc86796ec0e45b15df7d26392309d1bf7324
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/156303
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29447}
There are multiple places in webrtc code where alternative than
default rounding is desired. Typically this rounding is inlined.
e.g. as (<x> + <y>/2) / <y> making code more clumpsy (<y> might be long expression)
and unsafe for large values of <x>
This change introduce small helpers to address both concerns.
Bug: None
Change-Id: Icd8dcee80a697b7c50ba0b2e50295087d2be8670
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153354
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29222}
Add seek methods to FileWrapper, and refactor WavReader to use that
class instead.
Bug: webrtc:6463
Change-Id: Ifbb1989a072da6280ea5fc04b4beff991614dd53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147265
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28770}
