12 Commits

Author SHA1 Message Date
Taylor Brandstetter
b3c6810be3 Adding the ability to use a simulated clock for unit tests.
This will be useful for any tests that test objects with time-dependent
behavior. It will allow such tests to be written in such a way that their
outcome is more repeatable (less flaky), and will also allow such tests
to finish quicker. For example, a test for STUN timeout doesn't need to
wait the full timeout interval in real time; it can simply advance the
simulated clock.

BUG=webrtc:4925
R=pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/1895933003 .

Cr-Commit-Position: refs/heads/master@{#12950}
2016-05-27 21:16:07 +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
nisse
1bffc1d1a4 Rename rtc::Time64 --> rtc::TimeMillis.
In the discussion on https://codereview.webrtc.org/1888593004/, a more
decriptive name was suggested for Time64.

BUG=webrtc:5740

Review-Url: https://codereview.webrtc.org/1923213002
Cr-Commit-Position: refs/heads/master@{#12594}
2016-05-02 15:19:00 +00:00
nisse
b0c293c5ab Delete unused code in rtc timeutils.
BUG=webrtc:5740

Review URL: https://codereview.webrtc.org/1859413002

Cr-Commit-Position: refs/heads/master@{#12275}
2016-04-07 09:12:12 +00:00
honghaiz
34b11eb66e Using 64-bit timestamp to replace the 32-bit one in webrtc/p2p.
Also changed from unsigned to signed integer per the style guide.
By the way, I kept all delta-times to be 32-bit int.

The only things left in the p2p dir are
1. proberprober/main.cc where Time() is used as the input for a random number.
2. pseudotcp.cc: where 32-bit time info is sent over the wire.

BUG=webrtc:5636

Review URL: https://codereview.webrtc.org/1793553002

Cr-Commit-Position: refs/heads/master@{#12019}
2016-03-16 15:55:48 +00:00
sprang
1b3530b4df Make rtc::TimestampWrapAroundHandler handle backwards wrapping
Also fix a timestamp issue in video analyzer test.

BUG=webrtc:5637, webrtc:5537

Review URL: https://codereview.webrtc.org/1779773002

Cr-Commit-Position: refs/heads/master@{#11938}
2016-03-10 09:33:01 +00:00
Torbjorn Granlund
46c9cc0190 Provide method for returning certificate expiration time stamp.
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}
2015-12-01 12:06:46 +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
henrike@webrtc.org
99b4162ccf Rebase webrtc/base 6163:6216 (svn diff -r 6163:6216 http://webrtc.googlecode.com/svn/trunk/talk/base, apply diff manually)
BUG=3379
TBR=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6217 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-05-21 20:42:17 +00:00
henrike@webrtc.org
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
perkj@webrtc.org
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
henrike@webrtc.org
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