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22 Commits

Author SHA1 Message Date
pthatcher@webrtc.org
5d0071fb1f Build one of NSS or BoringSSL but not both. 
The libraries have some common symbols. When both are linked I observed NSS
SHA1_Update called followed by BoringSSL SHA1_Final, which results in a
segfault. We should only link one of these.

Based off of https://review.webrtc.org/25689004/

BUG=3855
R=pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31469004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7310 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-26 18:53:40 +00:00
pbos@webrtc.org
38344ed280 Move thread_annotations.h to webrtc/base/. 
R=andresp@webrtc.org, mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/27579004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7283 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-24 06:05:00 +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
c3c9015bc6 linux: remove stray libcrypto dependency 
Followup to CL 20049004, which looks like it added an unneeded -lcrypto
on linux.

BUG=3625
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/28459004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7168 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-12 16:11:38 +00:00
kjellander@webrtc.org
665d861115 Restore webrtc_base target until r7140 is rolled into Chromium. 
In r7140 the webrtc_base target was renamed to rtc_base. This
breaks our FYI bots for rolling WebRTC in Chromium's DEPS.
By re-adding a None target named webrtc_base, this transition
should be smoother.

TBR=henrikg@webrtc.org,
TESTED=Passed build/gyp_chromium on a Chromium checkout with src/third_party/webrtc replaced by a mount like this:
cd /path/to/chromium/src
sudo mount --bind /path/to/webrtc/trunk/webrtc third_party/webrtc

Review URL: https://webrtc-codereview.appspot.com/23589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7150 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-11 09:22:13 +00:00
tpsiaki@google.com
67eabc0938 Add schannel webrtc_base build using a new use_schannel gyp variable. 
R=henrike@webrtc.org, thorcarpenter@google.com

Review URL: https://webrtc-codereview.appspot.com/28409004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7141 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 18:06:47 +00:00
henrike@webrtc.org
b2efb6771c Put base tests in webrtc_tests.gyp 
BUG=N/A
R=andrew@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7140 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-09-10 17:28:19 +00:00
henrike@webrtc.org
66a3582170 Create a copy of talk/sound under webrtc/sound. 
BUG=3379
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22379004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6986 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-26 22:04:04 +00:00
henrike@webrtc.org
fb1eb43377 Rename linuxwindowpicker to x11windowpicker & only use it with use_x11 
These days we have Linux chromium builds that don't use X11. We don't
want webrtc to add an X11 dependency to those builds.

BUG=3625
R=henrike@webrtc.org, tnakamura@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20049004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6909 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-15 14:44:13 +00:00
henrike@webrtc.org
6ac22e6b47 Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 
R=andrew@webrtc.org, fbarchard@chromium.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/14029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6867 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-11 21:06:30 +00:00
henrike@webrtc.org
3763b9bda0 webrtc/base: removes linkage of crypto 
BUG=
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17069004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6853 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-08-07 21:26:18 +00:00
minyue@webrtc.org
74aaf29a0f Raw packet loss rate reported by RTP_RTCP module may vary too drastically over time. This CL is to add a filter to the value in VoE before lending it to audio coding module. 
The filter is an exponential filter borrowed from video coding module.

The method is written in a new class called PacketLossProtector (not sure if the name is nice), which can be used in the future for more sophisticated logic.

BUG=
R=henrika@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6709 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-16 21:28:26 +00:00
henrike@webrtc.org
5f2c81c17f webrtc/base: Fixes miss in base.gyp for windows. See https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle.gyp?r=6503#764 for the corresponding condition. 
BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/19849004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6615 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-07-07 17:42:45 +00:00
henrike@webrtc.org
4ddcc40d32 pkg-config-wrapper should not be run when build_nss is disabled (=0). 
BUG=b/15411893
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6538 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-25 20:14:13 +00:00
henrike@webrtc.org
a685c9df62 base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/ 
BUG=N/A
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20709004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6467 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-17 14:48:44 +00:00
henrike@webrtc.org
9f36c087f1 Makes it possible to prevent some third party libraries (jsoncpp and openssl) from being linked. This makes it possible to link webrtc with external implementations of those libraries in case the project depending on webrtc requires another version of those libraries. 
BUG=3379
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17699005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6455 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 21:35:20 +00:00
henrik.lundin@webrtc.org
1e3c5c248a Importing ThreadChecker class from Chromium 
The ThreadChecker class is imported/re-implemented from Chromium.
The implementation is changed to depend on WebRTC primitives.

R=andrew@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/21649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6446 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 11:34:44 +00:00
kjellander@webrtc.org
2bae3211b1 Add missing sources to webrtc/base/base.gyp 
During my work on setting up a GN build for
WebRTC, I discovered that the base.gyp is trying
to remove source files (for the Chromium build)
that are not added in the initial set.
I assume these files should be listed and that
GYP just doesn't complain when it's trying to
remove a file that is not present in the sources
list.

natserver_main.cc is also removed, since it's not used anywhere.

There are also a couple of other header files that are
used in other code that probably also should be listed in
base.gyp (please do this in another CL):
* compile_assert.h
* dscp.h
* move.h
* template_util.h

BUG=None
TEST=Trybots passing clobber compile step.
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6438 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-06-16 07:11:19 +00:00
tkchin@webrtc.org
7ca1edb31d Remove IOKit linkage from iOS builds. 
IOKit has been removed in iOS7, so link fails. iOS build succeeds after removing this setting and the corresponding one in talk/libjingle.gyp. Presubmit script tells me that CLs aren't allowed to touch both talk/ and webrtc/ at the same time so doing this separately.

BUG=
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/20509005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6191 4adac7df-926f-26a2-2b94-8c16560cd09d
 2014-05-19 21:05:10 +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