13 Commits

Author SHA1 Message Date
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
Erik Språng
1c3909899d Use rtc::time for all your timing needs!
Initial step of unifying so that base/timeutils.h and Clock/TimeTime
from system_wrappers use the same implementation.

BUG=webrtc:5463
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/1639543005 .

Cr-Commit-Position: refs/heads/master@{#11394}
2016-01-27 11:55:44 +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
henrikg
91d6edef35 Add RTC_ prefix to (D)CHECKs and related macros.
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.

Alternative solutions:
* Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable.
* Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce.
* Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable.
* Changes in Chromium for this is obviously not an option.

BUG=chromium:468375
NOTRY=true

Review URL: https://codereview.webrtc.org/1335923002

Cr-Commit-Position: refs/heads/master@{#9964}
2015-09-17 07:24:51 +00:00
andrew@webrtc.org
6ae5a6d7fe Add a target for the approved subset of rtc_base.
rtc_base drags in a bunch of unwieldly dependencies (e.g. nss and
json) not required for standalone webrtc (aka rtc/media). The root of
the problem appears to be that MessageQueue depends on a socket server.
(And since common.h -> logging.h -> thread.h -> messagequeue.h, this
dependency spreads quickly.)

This starts a new target for a "purified" subset of rtc_base. It adds
the files which are already being used, replacing the use of common.h
with checks.h. desktop_capture is a lost cause, and retains its
dependency on the full rtc_base.

The hope is that as additional components are desired they will be
cleaned and added to rtc_base_approved.

BUG=3806
R=andresp@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7188 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-16 01:03:29 +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