Commit Graph

14 Commits

Author SHA1 Message Date
1ee2125909 Adding PortAllocator option to support cases where sockets can't be bound.
This CL adds the flag "PORTALLOCATOR_ENABLE_ANY_ADDRESS_PORTS", which will
force the creation of ports not bound to any specific network interface.
These are normally only used when network enumeration fails or is disabled,
but in some circumstances (such as the one the test case adds), they're the
only thing that works.

This will result in extra ports being gathered, which is why it's only enabled
behind a flag for now. In the future, we could probably introduce more
sophisticated "pruning" logic that would lessen the impact of the extra ports
when they're redundant, and make the flag the default.

Some other minor changes that were required to make this use case work:

* Allow a TCPPort to be used for outgoing connections even if it tries and
  fails to create a server socket.
* Allow Bind to fail if being called before Connect, and the IP is an "any"
  address (0.0.0.0 or ::), since this bind would have been mostly pointless
  anyway.
* Prevent P2PTransprotChannel from keeping a "backup" candidate pair using
  an "any address" network; we only want this for actual networks.

BUG=webrtc:7798

Review-Url: https://codereview.webrtc.org/2936553003
Cr-Commit-Position: refs/heads/master@{#18578}
2017-06-13 22:49:45 +00:00
98e186c71c Remove VirtualSocketServer's dependency on PhysicalSocketServer.
The only thing the physical socket server was used for was
"Wait"/"WakeUp", but it could be replaced by a simple rtc::Event.

So, removing this dependency makes things less confusing; the fact that
VirtualSocketServer takes a PhysicalSocketServer may lead someone to
think it uses real sockets internally, when it doesn't.

BUG=None

Review-Url: https://codereview.webrtc.org/2883313003
Cr-Commit-Position: refs/heads/master@{#18172}
2017-05-17 01:00:06 +00:00
e077ee472a Deleted unused method EstimateMTU, and the WinPing class.
BUG=None

Review-Url: https://codereview.webrtc.org/2866183004
Cr-Commit-Position: refs/heads/master@{#18077}
2017-05-10 08:56:54 +00:00
37f5ecfd8f Replace NULL with nullptr or null in webrtc/base/.
BUG=webrtc:7147

Review-Url: https://codereview.webrtc.org/2718663005
Cr-Commit-Position: refs/heads/master@{#16878}
2017-02-27 22:06:41 +00:00
ede5da4960 Replace ASSERT by RTC_DCHECK in all non-test code.
Bulk of the changes were produced using

  git grep -l ' ASSERT(' | grep -v test | grep -v 'common\.h' |\
    xargs -n1 sed -i 's/ ASSERT(/ RTC_DCHECK(/'

followed by additional includes of base/checks.h in affected files,
and git cl format.

Also had to do some tweaks to #if !defined(NDEBUG) logic in the
taskrunner code (webrtc/base/task.cc, webrtc/base/taskparent.cc,
webrtc/base/taskparent.h, webrtc/base/taskrunner.cc), replaced to
consistently use RTC_DCHECK_IS_ON, and some of the checks needed
additional #if protection.

Test code was excluded, because it should probably use RTC_CHECK
rather than RTC_DCHECK.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2620303003
Cr-Commit-Position: refs/heads/master@{#16030}
2017-01-12 13:15:36 +00:00
9131efdb30 Read recv timestamps from socket (posix only).
This helps a lot on Android devices where the user threads can be scheduled with low priority when the app is in the background, causing spurious significantly delayed before a packet can be read from the socket. With this patch the timestamp is taken by the kernel when the packet actually arrives.

R=juberti@chromium.org
TBR=juberti@webrtc.org

BUG=webrtc:5773

Review URL: https://codereview.webrtc.org/1944683002 .

Cr-Commit-Position: refs/heads/master@{#12850}
2016-05-23 16:19:37 +00:00
555604a746 Replace scoped_ptr with unique_ptr in webrtc/base/
This propagated into various other places. Also had to #include headers that
were implicitly pulled by "scoped_ptr.h".

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1920043002

Cr-Commit-Position: refs/heads/master@{#12501}
2016-04-26 10:13:28 +00:00
5237aaf243 Convert usage of ARRAY_SIZE to arraysize.
ARRAY_SIZE is the old version of arraysize and does not cover
all the cases in C++, arraysize is a copy of Chromium's
version and thus have wider coverage.

BUG=None
R=tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1405023016

Cr-Commit-Position: refs/heads/master@{#10594}
2015-11-11 07:44:39 +00:00
0c4e06b4c6 Use suffixed {uint,int}{8,16,32,64}_t types.
Removes the use of uint8, etc. in favor of uint8_t.

BUG=webrtc:5024
R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1362503003 .

Cr-Commit-Position: refs/heads/master@{#10196}
2015-10-07 10:23:32 +00:00
c5d0d95fd8 Ensuring that UDP TURN servers are always used as STUN servers.
This was already working in most cases, but not for some corner cases:
* If the PORTALLOCATOR_ENABLE_SHARED_SOCKET flag is not set
* If both a STUN server and TURN server are configured

I added unit tests for these cases, and centralized the code that gets
STUN server addresses in order to fix these and any related issues.

BUG=webrtc:4215

Review URL: https://codereview.webrtc.org/1215713003

Cr-Commit-Position: refs/heads/master@{#9596}
2015-07-16 17:22:28 +00:00
67186fe00c Fix clang style warnings in webrtc/base
Mostly this consists of marking functions with override when
applicable, and moving function bodies from .h to .cc files.

Not inlining virtual functions with simple bodies such as

  { return false; }

strikes me as probably losing more in readability than we gain in
binary size and compilation time, but I guess it's just like any other
case where enabling a generally good warning forces us to write
slightly worse code in a couple of places.

BUG=163
R=kjellander@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47429004

Cr-Commit-Position: refs/heads/master@{#8656}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8656 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-03-09 22:24:25 +00:00
f048872e91 Adds a modified copy of talk/base to webrtc/base. It is the first step in
migrating talk/base to webrtc/base.

BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17479005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6129 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 18:00:26 +00:00
e9a604accd Revert 6107 "Adds a modified copy of talk/base to webrtc/base. I..."
This breaks Chromium FYI builds and prevent roll of webrtc/libjingle to Chrome.

http://chromegw.corp.google.com/i/chromium.webrtc.fyi/builders/Win%20Builder/builds/457


> Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
> 
> BUG=N/A
> R=andrew@webrtc.org, wu@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12199004

TBR=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14479004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6116 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-13 08:15:48 +00:00
2c7d1b39b9 Adds a modified copy of talk/base to webrtc/base. It is the first step in migrating talk/base to webrtc/base.
BUG=N/A
R=andrew@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6107 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-12 18:03:09 +00:00