16 Commits

Author SHA1 Message Date
nisse
7d2542623a Delete unneeded includes of base/common.h.
Bulk of the changes were done using

   git grep -l '#include "webrtc/base/common.h"' | \
     xargs sed -i '\,^#include.*webrtc/base/common\.h,d'

followed by adding back the include in the few places where it is
still needed, and in one case (pseudotcp.cc) instead deleting its use
of RTC_UNUSED.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2644103002
Cr-Commit-Position: refs/heads/master@{#16263}
2017-01-25 09:47:24 +00:00
sergeyu
d74886350e Fix PseudoTcp to handle incoming packets with invalid SEQ field
Previously PseudoTcp::process() didn't handle the case when the peer
sends a packet that's outside of the receive window, which was causing
DCHECK failures in the fuzzer.

BUG=681849

Review-Url: https://codereview.webrtc.org/2640173002
Cr-Commit-Position: refs/heads/master@{#16169}
2017-01-19 18:53:35 +00:00
nisse
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
nisse
c80e741ad0 Replace ASSERT(false) by RTC_NOTREACHED().
This cl was produced by

  git grep -l 'ASSERT(false)' |\
    xargs -n1 sed -i 's/ASSERT(false)/RTC_NOTREACHED()/'

followed by additional includes of base/checks.h in affected files,
git cl format to adjust spacing in webrtc/base/transformadapter.cc.
Finally, to make presubmit happy, one unnamed TODO marker was deleted
in that file.

This is a step towards deletion of base/common.h.

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2625003003
Cr-Commit-Position: refs/heads/master@{#16009}
2017-01-11 13:56:46 +00:00
sergeyu
d1523ca38f Fix header size check in PseudoTcp::parse().
BUG=chromium:620694
TBR=pbos@webrtc.org

Review-Url: https://codereview.webrtc.org/2073963002
Cr-Commit-Position: refs/heads/master@{#13177}
2016-06-17 01:08:19 +00:00
Honghai Zhang
82d7862fe7 Change default timestamp to 64 bits in all webrtc directories.
BUG=
R=pbos@webrtc.org, pthatcher@webrtc.org, solenberg@webrtc.org

Review URL: https://codereview.webrtc.org/1835053002 .

Cr-Commit-Position: refs/heads/master@{#12646}
2016-05-06 18:29:27 +00:00
kwiberg
3ec4679dd2 Replace scoped_ptr with unique_ptr in webrtc/p2p/
But keep #including scoped_ptr.h in .h files, so as not to break
WebRTC users who expect those .h files to give them rtc::scoped_ptr.

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1923163003

Cr-Commit-Position: refs/heads/master@{#12532}
2016-04-27 14:22:58 +00:00
jbauch
f1f87203d7 Split ByteBuffer into writer/reader objects.
This allows the reader to reference data, thus avoiding unnecessary
allocations and memory copies.

BUG=webrtc:5155,webrtc:5670

Review URL: https://codereview.webrtc.org/1821083002

Cr-Commit-Position: refs/heads/master@{#12160}
2016-03-30 13:43:44 +00:00
tfarina
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
Peter Boström
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
andresp@webrtc.org
ff689be3c0 Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/35079004

Cr-Commit-Position: refs/heads/master@{#8347}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8347 4adac7df-926f-26a2-2b94-8c16560cd09d
2015-02-12 11:55:32 +00:00
pthatcher@webrtc.org
e9db7fe80c Put pseudotcp back because remoting uses it.
R=andrew@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/36409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7887 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-13 01:56:39 +00:00
pthatcher@webrtc.org
dee76f3b89 Move the obvious/easy Jingle-specific code into webrtc/libjingle.
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7886 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-12-12 21:04:42 +00:00
henrike@webrtc.org
269fb4bc90 move xmpp and p2p to webrtc
Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and
webrtc/p2p. Also makes libjingle use those version instead of the one in the talk folder.

BUG=3379

Review URL: https://webrtc-codereview.appspot.com/26999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7549 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-28 22:20:11 +00:00
henrike@webrtc.org
28100cb388 Reverts r7459 "Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p."
BUG=N/A
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7472 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-17 22:03:39 +00:00
henrike@webrtc.org
d1ba6d9cbf Create a copy of talk/xmpp and talk/p2p under webrtc/libjingle/xmpp and webrtc/p2p.
BUG=3379
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/27709005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7459 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-15 17:30:28 +00:00