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wu@webrtc.org
de305014c6 Update talk to 55906045. 
Review URL: https://webrtc-codereview.appspot.com/3159005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5065 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 15:40:38 +00:00
turaj@webrtc.org
58cd31665c Address Clag Analyzer issues. 
Following are the issues related to NetEq 4, discovered by Clang Analyzer.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b44b95.html#EndPath
Valid; perhaps unlikely, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-6beef6.html#EndPath
Valid, addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2e3883.html#EndPath
Valid; Addressed

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-293659.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b875cd.html#EndPath
Valid; Addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/index.html
Not valid;

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-86f2ed.html#EndPath
Not Valid; the assert statement will be short-circuited, however I also added a check of nullity of |packet|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-3a5669.html#EndPath
Not Valid: |energy_input| and |energy_expand| are both non-negative, therefore if-statement condition on line 226 is not satisfied unless |energy_input| >= 1. Therefore |energy_input| cannot be zero after normalization to 14-bits, i.e. operations on lines 228 & 229.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2f914f.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-2332b1.html#EndPath
Valid; addressed.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-de8dea.html#EndPath
Not valid; |out_len| is set when Process() is called, however, it makes sense to initialize to zero when declaring |out_len|.

https://x20web.corp.google.com/~pbos/scan-build-2013-10-10-1/report-b671a3.html#EndPath
Valid; addressed.

BUG=
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2729005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5064 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 15:15:55 +00:00
asapersson@webrtc.org
7d6bd22019 Propagate estimated RTT from receivers to rtt observer. 
BUG=1613
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3119004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5063 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 12:14:34 +00:00
sprang@webrtc.org
da2c37b759 Video bandwidth not reported correctly 
ViEChannel::GetBandwidthUsage fails to aggregate video_bitrate_sent in
the same way as the total, fec and nack.

BUG=2579
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3199004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5062 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 09:49:03 +00:00
sergeyu@chromium.org
773e72797f Provide a MouseCursorMonitor::CreateForWindow implementation in *_null.cc 
Chromium issue:
https://code.google.com/p/chromium/issues/detail?id=310146

BUG=2551
R=wez@chromium.org

Review URL: https://webrtc-codereview.appspot.com/2759004

Patch from Daniel Nicoara <dnicoara@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5061 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 01:51:21 +00:00
wu@webrtc.org
de748c806c Remove unused make_scoped_ptr which causes an "ambiguous" error with chromium build. 
TEST=build
R=andrew@webrtc.org, fischman@webrtc.org
TBR=andrew

Review URL: https://webrtc-codereview.appspot.com/3149004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5059 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 20:43:27 +00:00
solenberg@webrtc.org
dce70ccb0b Add delay limit to ChokeFilter. 
BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5058 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 19:18:07 +00:00
wu@webrtc.org
f424cb8e13 Update talk to 55863981. 
TBR=mallinath

Review URL: https://webrtc-codereview.appspot.com/3089006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5056 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 17:57:33 +00:00
solenberg@webrtc.org
d6e46638ec Logging for BWE test framework. 
BUG=
R=stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5055 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 16:06:26 +00:00
wu@webrtc.org
cecfd1832d Update talk to 55821645. 
TEST=try bots
R=mallinath@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3139004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-30 05:18:12 +00:00
wu@webrtc.org
ec4cccc6b6 Update libyuv to 832. 
R=fbarchard@google.com, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3129004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5052 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 21:02:20 +00:00
pbos@webrtc.org
47ebbaddbb Make video/ only depend on video_engine_core. 
Fixes Android/Chromium build error. Previous dependencies included
VideoEngine tests that couldn't build on this configuration.

BUG=2535
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3109004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5050 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 13:11:56 +00:00
pbos@webrtc.org
def22b455b Stop DirectTransports in VideoSendStreamTests. 
Prevents racy packet delivery during or after Call destruction.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3099005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5049 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 10:12:10 +00:00
turaj@webrtc.org
55e1723713 Avoid a leak in AudioCodingModuleTest.TestIsac. The leak was caught by LSAN. 
BUG=2515
TEST=reproduced locally on linux and verified the fix resolves the issue.
R=henrik.lundin@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2499004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5048 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 04:40:09 +00:00
fischman@webrtc.org
9ca93a8b8e Explicitly @synthesize ObjC @properties 
This is required after https://code.google.com/p/gyp/source/detail?r=1768
turned on -Wobjc-missing-property-synthesis for ninja builds (until then it
was only enabled for xcode builds) to allow chromium_deps to roll in
webrtc/DEPS.

BUG=2560
R=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3089004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5047 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-29 00:14:15 +00:00
mikhal@webrtc.org
0aeb22e32c Adding tl0idx consideration for continuity 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2879004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5046 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 22:26:14 +00:00
pbos@webrtc.org
0803c03f9a Fix build/isolate.gypi path in webrtc_tests.gypi. 
BUG=2535
R=kjellander@webrtc.org
TBR=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3039005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5045 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 18:10:29 +00:00
fischman@webrtc.org
b7a171825b Drop ViEDecoderObserver::DecoderTiming impl now that WebRtcDecoderObserver rolled in r5038. 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3009004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5044 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 17:36:59 +00:00
pbos@webrtc.org
16e03b7bd8 Separate Call API/build files from video_engine/. 
BUG=2535
R=andrew@webrtc.org, mflodman@webrtc.org, niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2659004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5042 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 16:32:01 +00:00
pbos@webrtc.org
850bcbe855 Remove frame_callback.h include in webrtcvie.h. 
This file is about to be moved and it's not really needed. The class
I420FrameCallback is forward declared inside vie_image_process.h and
only used in talk/ for a no-op implementation that doesn't access the
pointer.

BUG=
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3039004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5041 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 15:41:17 +00:00
henrik.lundin@webrtc.org
1a3a6e5340 Removing the threshold from the auto-mute APIs 
The threshold is now set equal to the minimum bitrate of the
encoder. The test is also changed to have the REMB values
depend on the minimum bitrate from the encoder.

BUG=2436
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5040 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 10:16:14 +00:00
sprang@webrtc.org
fe5d36b6fe Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well. 
We will do some refactoring of video engine and would like to use the
same rtcp stats struct there. Both video and audio seem to use 8bit
fraction lost, so that is changed in the struct as well.

BUG=
R=henrik.lundin@webrtc.org, kjellander@webrtc.org, mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2959004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5039 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-28 09:21:07 +00:00
wu@webrtc.org
97077a3ab2 Update libjingle to 55618622. 
Update libyuv to r826.

TEST=try bots
R=niklas.enbom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2889004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 21:18:33 +00:00
fischman@webrtc.org
728bc0fa4c Add qiang.lu@intel.com to WATCHLISTS. 
(patch from http://review.webrtc.org/2859004/)

TBR=qiang.lu@intel.com

Review URL: https://webrtc-codereview.appspot.com/2989004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5037 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 19:20:02 +00:00
xians@webrtc.org
c94abd313e Use clang-format -style=chromium to correct the format in webrtc/modules/interface/module_common_types.h 
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2979004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5036 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 18:15:09 +00:00
wu@webrtc.org
e4e5683b41 Clean up tsan suppression file: 
1) remove suppressions that are already fixed.
2) merge duplicated suppressions.

TBR=mallinath
TEST=tsan try bot
BUG=1205,2078,2080

Review URL: https://webrtc-codereview.appspot.com/2949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5033 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 16:29:33 +00:00
xians@webrtc.org
0729460acb Added a "interleaved_" flag to webrtc::AudioFrame. 
And also did some format refactoring on the AudioFrame class, no change on the functionalities on those format refactoring code.

BUG=
TEST=compile
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5032 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-25 12:50:46 +00:00
vikasmarwaha@webrtc.org
442c5e47cd Update adapter.js to use TURN transport parameters for FF version 27 & above. 
R=juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2829004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5031 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 20:31:57 +00:00
vikasmarwaha@webrtc.org
d674a566d3 Update dc1 demo as it was using invalid data Constraint (Reliable:true) for SCTP. The constraint Reliable is not supported by Standard and ignored in our implementation. See issue 2511. 
R=dutton@google.com, jiayl@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2739004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5030 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 19:38:47 +00:00
andrew@webrtc.org
b3731da68f Prefix MOVE_ONLY_TYPE_FOR_CPP_03 with WEBRTC_. 
Will fix a redefinition error in Chromium against webrtc head.

TESTED=trybots
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5029 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 15:16:53 +00:00
henrik.lundin@webrtc.org
b56d0e383e Change the low-bitrate handling in BitrateControllerImpl 
Changing to using strategy classes rather than having two different
derived classes of BitrateControllerImpl. This enables run-time switching
of the strategy, which is now possible through a new API. The reason is
that it must fit the current design of ViE.

BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5028 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-24 09:24:06 +00:00
fischman@webrtc.org
37bb4974e7 Expose VideoCodingModule's decoder stats up the stack from VCMTiming to chrome://webrtc-internals. 
R=juberti@google.com, mikhal@webrtc.org, stefan@webrtc.org, wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2429004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5027 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 23:59:45 +00:00
wu@webrtc.org
d371a29227 Fix tsan failures for libjingle_unittest. 
1) Change AsyncSocket's SignalReadEvent and SignalWriteEvent's thread mode to multi_threaded_local as they can be accessed from different threads.
2) Protect NATServer::TransEntry::whitelist.
3) Protect PhysicalSocket:error_.

Detail failures can be seen from issue 2080, comment #5.

TBR=fischman@webrtc.org

RISK=P1
TEST=try bots and tsanv2
BUG=2080

Review URL: https://webrtc-codereview.appspot.com/2669005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5026 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 23:56:09 +00:00
andrew@webrtc.org
d1bcf1180a Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined. 
Works around a multiple definition error from webrtc and libjingle.

Corresponds to the libjingle change here:
https://critique.corp.google.com/#review/55489575-p10

TESTED=trybots
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2809004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5025 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 19:11:32 +00:00
andrew@webrtc.org
22858d4785 Add an extended filter option to audioproc. 
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5024 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 14:07:17 +00:00
asapersson@webrtc.org
042e91c2b2 Fix for incorrect RTT estimation. A too low RTT value could be estimated. 
R=andrew@webrtc.org, holmer@google.com, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2579005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5023 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 13:58:31 +00:00
henrik.lundin@webrtc.org
ba975e2078 Porting auto mute to new ViE API 
This CL also includes tests for the auto mute function. A few minor lint
warnings were fixed too. Note that the auto mute function is still work
in progress.

The callback ViEEncoderObserver::VideoAutoMuted was not ported from the
old API. This is TBD; see issue 2457.

BUG=2436
R=holmer@google.com, mflodman@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2340004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5021 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 11:04:57 +00:00
tina.legrand@webrtc.org
886aef09a8 Fixing broken tests in voe_auto_test extended 
This CL fixes the problem with voe_auto_test extended-codec test, as well as
extended-file test. First problem was that Opus was not added as a special case, like the other codecs, and the second problem was that the tests were not updated when test files were moved to the resources catalogue.

There are still some tests that fails. Here is a list of all extended tests and their status:

Base: fails - the reason seem to be that external transport has been removed.
CallReport: passes
Codec: passes (with this CL)
DTMF: passes
Encryption: fails or is dissabled?
VoEExternalMedia: passes
File: passes (with this CL)
Hardware: passes
NetEqStats: empty?
Network: passes
RTP_RTCP: fails
VideoSync: fails
VolumeControl: passes

BUG=issue2234
R=andrew@webrtc.org, henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2023004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5020 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-23 10:39:56 +00:00
wu@webrtc.org
8804a29951 Add CriticalSection to fakeaudiocapturemodule to protect the variables which will be accessed from process_thread_ and the main thread. 
TEST=try bots
BUG=1205
R=henrike@webrtc.org, kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2419004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5019 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 23:09:20 +00:00
wu@webrtc.org
4d7116be7a Fix tsan failures on filevideocapturer.cc. 
1) init start_time_ns_ before the file_read_thread_ is started to avoid data racing as the start_time_ns_ will also be read by the file_read_thread_.
2) add CriticalSection to protect |finished_| that is accessed by FileReadThread and the main thread

Also remove the suppression for filemediaengine.cc, which has already been fixed in other cl.

TBR=henrike@webrtc.org
TEST=try bots and manual tsan v2 test
BUG=2078

Review URL: https://webrtc-codereview.appspot.com/2509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5018 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 18:41:17 +00:00
vikasmarwaha@webrtc.org
90d8719fd7 Radix should be specified when calling ParseInt function in adapter.js. Refer to issue 2490. 
R=dutton@google.com, juberti@google.com

Review URL: https://webrtc-codereview.appspot.com/2709006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5017 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 18:02:41 +00:00
kjellander@webrtc.org
8575980e16 Add default trybots for WebRTC try server. 
Today, our tryjobs default to run on all trybots since
we don't have any default list configured in PRESUBMIT.py.
Because of this, the --testfilter argument doesn't work
unless you also specify --bot when sending the tryjob.

With this CL, it is possible to use --testfilter without
--bot.
It also gets the benefit of excluding unnecessary bots
when doing platform-specific changes.

Most of the code is copied from Chromium's src/PRESUBMIT.py:
https://code.google.com/p/chromium/codesearch#chromium/src/PRESUBMIT.py&l=1030

TEST=tested submitting a tryjob with git try -t compile.
BUG=none
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2619004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5016 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 16:47:40 +00:00
andrew@webrtc.org
31628aae7e Upgrade scoped_ptr to Chromium's latest version. 
Analogous to the recent libjingle change: http://cl/54929753-p10.
This supports scoped_ptr<T[]> and scoped_ptr<C, FreeDeleter> rather
than scoped_array and scoped_ptr_malloc respectively.

- Add Chromium's template-based COMPILE_ASSERT. We didn't have this
previously in order to support the macro in C. Instead, move the
existing macro to compile_assert_c.h.
- Additionally copy the move.h and template_util.h depedencies and add
the WARN_UNUSED_RESULT macro.
- Leave scoped_array and scoped_ptr_malloc for now, but mark as
deprecated.
- Remove scoped_ptr foo(NULL) use. The default constructor handles it.
- Remove the now redundant COMPILE_ASSERT from peerconnection_jni.cc.
- Add a CHECK_ARRAY_SIZE macro to rtp_format_vp8_unittest.cc to remove
some repeated code.

TESTED=trybots
R=pbos@webrtc.org, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2449005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5015 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 12:50:00 +00:00
kjellander@webrtc.org
06b60c07b7 Roll chromium_revision 228675:229708 
This will pick up the -Wunused-const-variable
Clang warning being enabled by default (chromium:307668).

BUG=none
TEST=trybots passing.
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2629004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5014 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 12:09:48 +00:00
andrew@webrtc.org
621df678c8 WEBRTC_{BIG, LITTLE}_ENDIAN -> WEBRTC_ARCH_{BIG, LITTLE}_ENDIAN. 
Mostly to remove a long-standing TODO...

TESTED=trybots
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2369005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5013 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 10:27:23 +00:00
marpan@webrtc.org
943e3b95a6 Add CurrentLayerId() to temporal layers. 
same patch as: https://webrtc-codereview.appspot.com/2427004/

TBR=holmer@google.com

Review URL: https://webrtc-codereview.appspot.com/2729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5012 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-22 01:55:07 +00:00
mallinath@webrtc.org
50bc553852 Reenable DTLS renegotiation unittest in libjingle. 
This test is failing on memcheck bots. After investigation problem per
say is not in this particular unittest and rather is in suite. So this test
is added to memcheck exclude list.

R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2689004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5011 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 17:58:35 +00:00
elham@webrtc.org
9c735c4e25 Updated WebRTC version to 3.45 
R=wu@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2669004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5009 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 16:34:50 +00:00
solenberg@webrtc.org
8215106371 Framework for testing bandwidth estimation. 
BUG=
R=mflodman@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2317004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5008 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 14:23:26 +00:00
henrik.lundin@webrtc.org
29dd0de5b3 Changing the bitrate clamping in BitrateControllerImpl 
This CL implements an alternative to the bitrate clamping that is done
in BitrateControllerImpl. The default behavior is unchanged, but if
the new algorithm is enabled the behavior is as follows:
When the new bitrate is lower than the sum of min bitrates, the
algorithm will give each observer up to its min bitrate, one
observer at a time, until the bitrate budget is depleted. Thus,
with this change, some observers may get less than their min bitrate,
or even zero.

Unit tests are implemented.

Also fixing two old lint warnings in the affected files.

This change is related to the auto-muter feature.

BUG=2436
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2439005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5007 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 14:00:01 +00:00