1. It signals network changed events whenever there are more than one IP address in a network.
2. It does not signal network changed events if a network disconnects and connects again.
Also changed DumpNetworks for better debugging.
BUG=webrtc:5096
Review URL: https://codereview.webrtc.org/1421433003
Cr-Commit-Position: refs/heads/master@{#11107}
This patch removes StringToIP() methods as fixes the TODO there and
there are no callers at the moment for these methods.
BUG=None
R=perkj@webrtc.org
Review URL: https://codereview.webrtc.org/1535993002
Cr-Commit-Position: refs/heads/master@{#11088}
Incorrect argument order, also added unittest which should've been there
in the first place.
Also renames AtomicLoadPtr to AcquireLoadPtr to match non-ptr version.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1537923003 .
Cr-Commit-Position: refs/heads/master@{#11086}
There is an issue in PhysicalSocket::Accept where the flag to continue
listening is not set in "enabled_events_" if "accept" returns an error.
This CL fixes this (initial idea by silviu.cpp@gmail.com).
BUG=webrtc:2030
Review URL: https://codereview.webrtc.org/1452903006
Cr-Commit-Position: refs/heads/master@{#11080}
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
This makes it clearer that the IP address has been stripped.
BUG=chromium:254329
Review URL: https://codereview.webrtc.org/1516163003
Cr-Commit-Position: refs/heads/master@{#11005}
rtc::PlatformThreadId is pid_t (32-bit signed int) on Linux and Mac,
but DWORD (32-bit unsigned int) on Windows.
Using the %d printf specifier is therefore not correct on Windows,
and Clang would warn about it:
..\..\third_party\webrtc\base\event_tracer.cc(124,46) : error: format specifies
type 'int' but the argument has type 'rtc::PlatformThreadId' (aka 'unsigned
long') [-Werror,-Wformat]
e.phase, e.timestamp, e.pid, e.tid);
^~~~~
This commit fixes the problem by explicitly casting to int before printing.
BUG=82385
Review URL: https://codereview.webrtc.org/1514253002 .
Cr-Commit-Position: refs/heads/master@{#10982}
Fixes one sign mismatch warning, and one "const has no effect and is
ignored" warning.
BUG=chromium:567877
Review URL: https://codereview.webrtc.org/1510233002
Cr-Commit-Position: refs/heads/master@{#10976}
Created a simple unit test for the new random number generator. (It mostly tests
that the generated numbers are consistent with the intended distribution, e.g. uniform.
It is not a comprehensive test of the quality of the random numbers.)
Several assertions in OveruseDetectorTest seem to depend on the exact sequence of random numbers. I updated those numbers to work with the new PRNG.
Compute the standard deviation of the expected result in TestReorderFilter instead of passing an uncertainty parameter.
BUG=webrtc:5177
Review URL: https://codereview.webrtc.org/1457023002
Cr-Commit-Position: refs/heads/master@{#10965}
Reason for revert:
Broke downstream compile step, possibly relandable when using a MSVC version that has constexpr, other than that I'm out of ideas.
.../webrtc/base/atomicops.h:71:8: note: no known conversion for argument 1 from '<brace-enclosed initializer list>' to 'const rtc::AtomicInt&'
Original issue's description:
> Reland of "Create rtc::AtomicInt POD struct."
>
> Relands https://codereview.webrtc.org/1420043008/ with brace initializers
> instead of constructors hoping that they won't introduce static
> initializers.
>
> BUG=
> R=tommi@webrtc.org
>
> Committed: https://crrev.com/84f0970d100e67a1dc4fe9a1b16b7d293302044e
> Cr-Commit-Position: refs/heads/master@{#10920}
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=
Review URL: https://codereview.webrtc.org/1505053002
Cr-Commit-Position: refs/heads/master@{#10922}
Logs tracing events (TRACE_EVENT0 and friends) to storage in a format
compatible with chrome://tracing which can be used for performance
evaluation, finding lock contention and other sweet things). Tracing is
still basic and doesn't contain thread metadata or logging of tracing
arguments.
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1457383002 .
Cr-Commit-Position: refs/heads/master@{#10921}
We convert ASN1 time via std::tm to int64_t representing milliseconds-since-epoch. We do not use time_t since that cannot store milliseconds, and expires for 32-bit platforms in 2038 also for seconds.
Conversion via std::tm might might seem silly, but actually doesn't add any complexity.
One would expect tm -> seconds-since-epoch to already exist on the standard library. There is mktime, but it uses localtime (and sets an environment variable, and has the 2038 problem).
The ASN1 TIME parsing is limited to what is required by RFC 5280.
BUG=webrtc:5150
R=hbos@webrtc.org, nisse@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1468273004 .
Cr-Commit-Position: refs/heads/master@{#10854}
The callback keeps a reference to an object until the callback goes out of scope.
Review URL: https://codereview.webrtc.org/1487493002
Cr-Commit-Position: refs/heads/master@{#10847}
Also changes presubmit script to not run cpplint on objc dirs.
BUG=
Review URL: https://codereview.webrtc.org/1467173006
Cr-Commit-Position: refs/heads/master@{#10815}
* Move PlatformThread to rtc::.
* Remove ::CreateThread factory method.
* Make non-scoped_ptr from a lot of invocations.
* Make Start/Stop void.
* Remove rtc::Thread priorities, which were unused and would collide.
* Add ::IsRunning() to PlatformThread.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1476453002 .
Cr-Commit-Position: refs/heads/master@{#10812}
The corresponding set of overrides weren't moved when logging.cc etc.
was moved over. This wasn't noticed because all existing targets before
webrtc fuzzers used to link both rtc_base and rtc_base_approved in
Chromium. Also adding //base:base as a dependency, this used to be
linked in by other targets either way before but generated build errors
when a target solely depends on rtc_base_approved.
BUG=webrtc:4771
R=kjellander@webrtc.orgTBR=henrikg@webrtc.org
Review URL: https://codereview.webrtc.org/1473223005 .
Cr-Commit-Position: refs/heads/master@{#10792}
(patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
Relanding after fixing CallAndModifyStream to account for new
procedures for adding/removing a track from a stream.
Original issue's description:
> Adding the ability to create an RtpSender without a track.
>
> This CL also changes AddStream to immediately create a sender, rather
> than waiting until the track is seen in SDP. And the PeerConnection now
> builds the list of "send streams" from the list of senders, rather than
> the collection of local media streams.
>
> Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> Cr-Commit-Position: refs/heads/master@{#10414}
Review URL: https://codereview.webrtc.org/1468113002
Cr-Commit-Position: refs/heads/master@{#10790}
With this in, the only compilation errors left seems
related to yasm and libjpeg_turbo.
Notice the below example builds x86 builds (not ARM) since if
specifying target_cpu="arm", the gn step fails (separate issue).
BUG=webrtc:5213, webrtc:5195, chromium:459705
TESTED=Passing compilation with:
gn gen --args="target_os=\"ios\"" out/Default
ninja -C out/Default rtc_base audio_device
Review URL: https://codereview.webrtc.org/1471663002
Cr-Commit-Position: refs/heads/master@{#10763}
Also removes all virtual methods. Permits using a thread from
rtc_base_approved (namely event tracing).
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1469013002
Cr-Commit-Position: refs/heads/master@{#10760}
We don't need it anymore now that we can use std::vector::data().
Review URL: https://codereview.webrtc.org/1470843003
Cr-Commit-Position: refs/heads/master@{#10755}
Reason for revert:
Still breaking CallAndModifyStream. Chromium CL intended to fix it (https://codereview.chromium.org/1435713002/) wasn't sufficient, because I forgot to call addStream/removeStream on the second connection.
Original issue's description:
> Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
>
> Reason for revert:
> Relanding, after changing the expectations of WebRtcBrowserTest.CallAndModifyStream.
>
> Original issue's description:
> > Revert of Adding the ability to create an RtpSender without a track. (patchset #8 id:140001 of https://codereview.webrtc.org/1413713003/ )
> >
> > Reason for revert:
> > Causing a compiler warning, and causing WebRtcBrowserTest.CallAndModifyStream to fail.
> >
> > Original issue's description:
> > > Adding the ability to create an RtpSender without a track.
> > >
> > > This CL also changes AddStream to immediately create a sender, rather
> > > than waiting until the track is seen in SDP. And the PeerConnection now
> > > builds the list of "send streams" from the list of senders, rather than
> > > the collection of local media streams.
> > >
> > > Committed: https://crrev.com/ac9d92ccbe2b29590c53f702e11dc625820480d5
> > > Cr-Commit-Position: refs/heads/master@{#10414}
> >
> > TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> >
> > Committed: https://crrev.com/8f46c63f6f764254892f4111b54aa1cc8f32eeeb
> > Cr-Commit-Position: refs/heads/master@{#10417}
>
> TBR=pthatcher@webrtc.org,pthatcher@chromium.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
>
> Committed: https://crrev.com/6834fa10f142bf5e2275142acb834898911d09ae
> Cr-Commit-Position: refs/heads/master@{#10730}
TBR=pthatcher@webrtc.org,pthatcher@chromium.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
Review URL: https://codereview.webrtc.org/1460323002
Cr-Commit-Position: refs/heads/master@{#10732}
