admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
4,063 Commits 1 Branch 0 Tags
Commit Graph

54 Commits

Author SHA1 Message Date
pbos@webrtc.org
4ca7d3f9fe Replace MapWrapper with std::map<>. 
MapWrapper was needed on some platforms where STL wasn't supported, we
now use std::map<> directly.

BUG=2164
TEST=trybots
R=henrike@webrtc.org, phoglund@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2001004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4530 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-12 19:51:57 +00:00
pbos@webrtc.org
12dc1a38ca Switch C++-style C headers with their C equivalents. 
The C++ headers define the C functions within the std:: namespace, but
we mainly don't use the std:: namespace for C functions. Therefore we
should include the C headers.

BUG=1833
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1917004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-05 16:22:53 +00:00
pbos@webrtc.org
a2a2718a6c Fix some chromium-style warnings in webrtc/system_wrappers/ 
BUG=163
R=henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1906004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4464 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-08-01 17:26:15 +00:00
pbos@webrtc.org
d900e8bea8 Proper spacing for end-of-namespace comments. 
BUG=
R=mflodman@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1760006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4293 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-07-03 15:12:26 +00:00
fischman@webrtc.org
1d4a2d5daf Move TickTime::QueryOsForTicks out-of-line 
This inline function is no longer expanded on arm Android, but on x86 Android it
will still be expanded. Move it out-of-line to make things consistent.

This change list will also fix a potential bug on webrtc for Android:
Since the inline function won't be expanded on arm Android,
TickTime::MillisecondTimestamp and Clock::GetRealTimeClock()->TimeInMilliseconds
will be treated as function call, due to macro WEBRTC_CLOCK_TYPE_REALTIME's
guard defined in system_wrappers module they will get current time using
CLOCK_REALTIME.

But on x86 Android, the inline function will be expanded to where it's been
called, if the call happens in other compilation units which don't have
WEBRTC_CLOCK_TYPE_REALTIME definition, it will get current time using
CLOCK_MONOTONIC, while Clock::GetRealTimeClock()->TimeInMilliseconds will always
use CLOCK_REALTIME, then there will be two types of time in x86 Android which
will cause some weird issues like all received remote streams will be dropped
due to future render timestamp.

BUG=None
TEST=WebRTCViEDemo application works well on both arm and x86 Android
R=fischman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1688004

Patch from Jeremy Mao <yujie.mao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4274 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-27 17:15:20 +00:00
andrew@webrtc.org
c1eb560a5c Replace the old resampler with SincResampler in the voice engine signal path. 
* The old resampler was found to have a wraparound bug.
* Remove support for the old resampler from PushResampler.
* Use PushResampler in AudioCodingModule.
* The old resampler must still be removed from the file utility.

BUG=webrtc:1867,webrtc:827
TESTED=unit tests, Chrome using apprtc and voe_cmd_test to verify wrap-around is corrected, voe_cmd_test running through all supported codec sample rates and channels to verify good quality audio
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1590004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4156 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-06-03 19:00:29 +00:00
pbos@webrtc.org
acaf3a1b13 Include files from webrtc/.. paths in system_wrappers/ 
BUG=1662
R=henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1550004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4111 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-27 15:07:45 +00:00
niklas.enbom@webrtc.org
3be565b502 Refactoring for typing detection 
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1370004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3976 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-07 21:04:24 +00:00
andrew@webrtc.org
f9c289bafe Consolidate all third party licenses in LICENSE_THIRD_PARTY. 
* Add the full license to all third party files.
* Correct some entries in LICENSE_THIRD_PARTY which were missing the full
license.
* Relicense all Chromium-licensed files under WebRTC.
* Remove third_party_mods/, which is now redundant.

R=jan.linden@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1396004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3959 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-05-05 18:54:10 +00:00
stefan@webrtc.org
b8e7f4cc97 Change capture interface to use NTP capture time. 
Move NTP functionality to Clock.

BUG=1563
TEST=trybots and vie_auto_test --automated

Review URL: https://webrtc-codereview.appspot.com/1313005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3842 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-12 11:56:23 +00:00
stefan@webrtc.org
7da3459b2a Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." 
This reverts commit 4954b3650192d78037714138a5c519ef08f2670e.
Reverts r3799

TBR=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1308004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 14:56:29 +00:00
stefan@webrtc.org
afcc6101d0 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. 
We should consider making the same change to the render timestamps generated at the receiver.

BUG=1563

Review URL: https://webrtc-codereview.appspot.com/1283005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 13:37:40 +00:00
pbos@webrtc.org
046deb9b20 WebRtc_Word32 -> int32_t in system_wrappers 
BUG=314

Review URL: https://webrtc-codereview.appspot.com/1301004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3791 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-04-09 09:06:11 +00:00
henrike@webrtc.org
93bea51517 Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
Recommitting https://code.google.com/p/webrtc/source/detail?r=3736 after fixing build break.

BUG=8404677
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3739 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-28 15:58:49 +00:00
wu@webrtc.org
80fccc29de Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..." 
> Removed CPU APIs from VoEHardware. Code is now only used by test applications.
> 
> BUG=8404677
> 
> Review URL: https://webrtc-codereview.appspot.com/1238004

TBR=henrike@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/1267004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3737 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 23:38:21 +00:00
henrike@webrtc.org
4c138e8fca Removed CPU APIs from VoEHardware. Code is now only used by test applications. 
BUG=8404677

Review URL: https://webrtc-codereview.appspot.com/1238004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3736 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-27 21:23:42 +00:00
andrew@webrtc.org
c1ffd337f1 Add trace printouts to all unit tests. 
Unfortunately, this requires splitting system_wrappers_unittests out of system_wrappers.gyp to avoid a cyclic dependency.

TESTED=ran a few unit tests and observed printouts

Review URL: https://webrtc-codereview.appspot.com/1221006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3711 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-22 17:13:23 +00:00
kma@webrtc.org
a0936a6e45 Limit ARM instruction "strheq" to Apple's clang compiler only. 
bug =
Review URL: https://webrtc-codereview.appspot.com/1111008

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3583 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-03-01 00:20:10 +00:00
kma@webrtc.org
2f9bd247ad Ported assembly coding in APM from Android to iOS. 
Bugs=none
Test=trybots, and offline file bit-exact tests.
Review URL: https://webrtc-codereview.appspot.com/1066009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3563 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-23 04:16:59 +00:00
wjia@webrtc.org
7bf7326d0b Remove WEBRTC_TRACE completely when tracing is disabled. 
This will help to cut the code size since those logging messages are removed.
Contributed by Henrik Ellner.
Review URL: https://webrtc-codereview.appspot.com/1125004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3560 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-22 00:54:58 +00:00
andrew@webrtc.org
076fc12539 Modify SincResampler to build in webrtc. 
This is the first in a series of CLs to bring arbitrary resampling to webrtc.

* Replace Chromium-specific helpers with their respective webrtc versions.
* Add a second constructor to permit runtime selection of block_size.
* Add stringize_macros to system_wrappers.

BUG=webrtc:1395
TESTED=unit tests

Review URL: https://webrtc-codereview.appspot.com/1097012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3518 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-15 03:54:22 +00:00
andrew@webrtc.org
45eab19e7d Import stringize_macros from Chromium. 
Committing the originals to make further reviews cleaner.

TBR=bjornv
BUG=webrtc:1395

Review URL: https://webrtc-codereview.appspot.com/1106005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3509 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-02-13 23:37:14 +00:00
kma@webrtc.org
16d540eff1 Fixed text relocation code related to ARM assembly code. 
Refer to WebRTC issue 1300.
Review URL: https://webrtc-codereview.appspot.com/1055004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3409 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-25 03:18:05 +00:00
kma@webrtc.org
4782911572 Optimized WebRtcIsacfix_Time2Spec() for iSAC-Fix in ARM Neon processor. 
Review URL: https://webrtc-codereview.appspot.com/1005004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3404 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-24 01:37:33 +00:00
stefan@webrtc.org
a678a3baee Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. 
TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1044004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-21 07:42:11 +00:00
henrike@webrtc.org
a3e6bec23a Posix Thread: Removes the setting of the run function to NULL which could cause data race. 
BUG=http://code.google.com/p/chromium/issues/detail?id=103711
TESTED=Code analysis (no tools)

Review URL: https://webrtc-codereview.appspot.com/1008006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3388 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-18 16:39:21 +00:00
stefan@webrtc.org
20ed36dada Break out RtpClock to system_wrappers and make it more generic. 
The goal with this new clock interface is to have something which is used all
over WebRTC to make it easier to switch clock implementation depending on where
the components are used. This is a first step in that direction.

Next steps will be to, step by step, move all modules, video engine and voice
engine over to the new interface, effectively deprecating the old clock
interfaces. Long-term my vision is that we should be able to deprecate the clock
of WebRTC and rely on the user providing the implementation.

TEST=vie_auto_test, rtp_rtcp_unittests, trybots

Review URL: https://webrtc-codereview.appspot.com/1041004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-17 14:01:20 +00:00
phoglund@webrtc.org
5c8d9d30e2 Reformatted tick_util. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1014004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3330 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-03 09:50:17 +00:00
phoglund@webrtc.org
daabfd25a6 Reformatted trace* files. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1015004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3329 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-03 09:37:03 +00:00
phoglund@webrtc.org
ec9c942e45 Reformatted thread and static_instance. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1006005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3324 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-01-02 08:45:03 +00:00
phoglund@webrtc.org
6bc5d4dc07 Reformatted sort. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/998006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3316 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-19 14:55:24 +00:00
phoglund@webrtc.org
6e0ce73741 Reformatted map classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1006004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3308 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 17:18:35 +00:00
phoglund@webrtc.org
59ad541e57 Reformatted rw_lock classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/1007004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3305 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-18 15:20:35 +00:00
hclam@chromium.org
c5fcb0879b Update trace_event.h to match the one in Chromium 
Chromium's trace_event.h has updated to remove some not-well-used features.
Update WebRTC's copy to match.
Review URL: https://webrtc-codereview.appspot.com/995006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3296 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 21:16:46 +00:00
phoglund@webrtc.org
52d981f60c Reformatted list classes. 
BUG=
TEST=Trybots

Review URL: https://webrtc-codereview.appspot.com/995004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3291 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-14 09:52:34 +00:00
kma@webrtc.org
7877b0f6d2 Added noexecstack markers for assembly files (webrtc issue 1172). 
Webrtc builds on ios, linux, android and other major platforms passed. Didn't do chrome build test.
Review URL: https://webrtc-codereview.appspot.com/987004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3275 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 23:22:13 +00:00
phoglund@webrtc.org
740be44af5 Reformatted file_* classes. 
BUG=
TEST=Trybots.

Review URL: https://webrtc-codereview.appspot.com/980004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3268 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 12:52:15 +00:00
hclam@chromium.org
4e16f25774 Remove atomicops.h from WebRTC 
atomicops.h are not necessary in trace_event.h similar to the port in WebKit.
It will cause a benign race condition detected by TSAN. If it shows up in
TSAN we will either suppress it or annotate it with dynamic annotations.

BUG=1215
Review URL: https://webrtc-codereview.appspot.com/982004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3267 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-12 01:13:19 +00:00
hclam@chromium.org
770a01e3b0 Fix build by including trace_event_internal in webrtc namespace 
TBR=ajm@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/969017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3265 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 22:51:38 +00:00
hclam@chromium.org
ad7efa6944 Port Chromium's trace_event.h to WebKit and add 
trace_event.h is ported from Chromium code.

These files are defined new for WebRTC:
* event_tracer.h
* event_tracer.cc
* event_tracer_unittest.cc
Review URL: https://webrtc-codereview.appspot.com/933034

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3262 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-11 21:19:08 +00:00
phoglund@webrtc.org
5bbe069f28 Reformatted event* classes. 
TEST=Ran trybots.

Review URL: https://webrtc-codereview.appspot.com/972012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3253 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-10 10:44:37 +00:00
hclam@chromium.org
81cffd1f2c Port Chromium's atomicops to WebRTC 
Porting Chromium's base/atomicops.h and friends into WebRTC.

Included the original unit test in Chromium.
Review URL: https://webrtc-codereview.appspot.com/964026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3241 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-05 22:51:01 +00:00
phoglund@webrtc.org
ba21c95e15 Reformatted data_log. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/974007

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3221 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-12-03 09:01:21 +00:00
phoglund@webrtc.org
99f7c917d2 Reformatted critical_section wrappers. 
BUG=
TEST=ran trybots

Review URL: https://webrtc-codereview.appspot.com/971012

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3210 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-30 10:44:49 +00:00
andrew@webrtc.org
c3e5d3422b Add a logging_no_op.cc when enable_tracing==0. 
This should hopefully fix static initializer warnings when rolling webrtc
in Chromium.

TEST=logging_unittest succeeds with enable_tracing==1 and fails appropriately with enable_tracing==0.

Review URL: https://webrtc-codereview.appspot.com/939026

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3159 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-23 19:30:59 +00:00
phoglund@webrtc.org
b15d285458 Reformatted most of the CPU stuff in system_wrappers. 
BUG=

Review URL: https://webrtc-codereview.appspot.com/930025

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3142 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-21 08:02:57 +00:00
andrew@webrtc.org
655d8f56f6 Add a kTraceTerseInfo level for non-verbose logging. 
Review URL: https://webrtc-codereview.appspot.com/937023

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3134 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-20 07:34:45 +00:00
kma@webrtc.org
55cd78cfc2 Porting ARM optimization from Android to ios. 
Tested APM and iSAC in Android. Bit-exact with original versions.
Changes include removing or changing some GCC derivatives (e.g. .fnstart, .hword), instruction syntax, etc.
Review URL: https://webrtc-codereview.appspot.com/934009

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3124 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-17 00:22:46 +00:00
andrew@webrtc.org
50419b0777 Add libjingle-style stream-style logging. 
Add a highly stripped-down version of libjingle's base/logging.h. It is
a thin wrapper around WEBRTC_TRACE, maintaining the libjingle log
semantics to ease a transition to that format.

Also add some helper macros for easy API and function failure logging.

Review URL: https://webrtc-codereview.appspot.com/931010

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3099 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 19:07:54 +00:00
phoglund@webrtc.org
a36d75a03c Reformatted condition_variable* in system_wrappers. 
BUG=
TEST=Ran trybots.

Review URL: https://webrtc-codereview.appspot.com/937017

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3095 4adac7df-926f-26a2-2b94-8c16560cd09d
 2012-11-14 09:55:04 +00:00