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1427 Commits

Author SHA1 Message Date
marpan@webrtc.org
bde3056567 Fix for video_processor_intergration_tests to run in parallel. 
BUG=2601.
R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3519004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5091 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 20:59:29 +00:00
kjellander@webrtc.org
c4225b63bb Update getUserMedia W3C conformance tests. 
This CL updates these tests to the spec as of
http://dev.w3.org/2011/webrtc/editor/archives/20130824/getusermedia.html

There are still a lot of functionality that lacks testing. I've put a bunch of TODOs in there but I'm unlikely to get time to implement them all any time soon...

TEST=local testing with Chrome Canary.
BUG=
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2639004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5090 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 13:26:34 +00:00
asapersson@webrtc.org
8bad50e845 Sending status fix for module. 
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5089 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-06 10:45:58 +00:00
kjellander@webrtc.org
7a36cb408b Add missing dependencies to .isolate files 
Also fix invalid paths in video_engine_tests.isolate.

TEST=trybots passing compile step (no .isolate use is deployed on them yet)
BUG=chromium:300017
R=pbos@webrtc.org
TBR=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3399005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5084 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-05 14:28:57 +00:00
fischman@webrtc.org
b8cb85b348 Fix broken build on x86 Android 
BUG=2545
R=fischman@webrtc.org, henrike@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3019004

Patch from Lu Quiang <qiang.lu@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5081 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 19:06:08 +00:00
asapersson@webrtc.org
766154aa1d Removed unused code. 
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3219004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5073 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-04 08:35:50 +00:00
kjellander@webrtc.org
e2df8b7f01 Make video quality analysis unittests print to log instead of stdout. 
I think it's best to avoid printing these perf numbers since
when we turn on perf measurements for Android, it will be for
all tests as far as I understand it works today.

TEST=trybots passing tools_unittests
BUG=none
R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3109005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5072 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-11-03 18:34:51 +00:00
sheu@chromium.org
5dd2ecb32d Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit f4ca3808bd9ec2293ec205f2f4a7d9739ce1f2df.

TBR=niklas.emblom@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/3269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5071 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 23:41:04 +00:00
sheu@chromium.org
74e6e8458e Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
BUG=chromium:310271
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
TBR=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3239005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5070 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:48:16 +00:00
sheu@chromium.org
d705649edf Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420" 
This reverts commit 99f9743fe39066ba93b41f2b0a417696cbbd06fb.

Revert while build breakage is fixed.

BUG=None
TBR=niklas.emblom@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5069 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 21:20:15 +00:00
sheu@chromium.org
1a4ed0d70c Remove extra copy in VideoCaptureImpl::IncomingFrameI420 
Add support for aliasing a I420VideoFrame (and internally, a Plane) to an
existing memory buffer without taking ownership.  Use this to remove an extra
copy in VideoCaptureImpl::IncomingFrameI420.

BUG=1128
TEST=local build, run Chromium on ARM, build, run Chromium/unittests on Linux
R=fischman@webrtc.org, mflodman@webrtc.org, mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3179004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5068 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-31 20:32:28 +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
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
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
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
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
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
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
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
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
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
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
henrik.lundin@webrtc.org
0d19ed9a06 AutoMute: Adding channel_id parameter to callback. 
BUG=2436
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2390004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5006 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 12:37:13 +00:00
pbos@webrtc.org
fe1ef935e7 Implement I420FrameCallbacks in Call. 
BUG=2425
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2393004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5005 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 10:34:43 +00:00
pbos@webrtc.org
e05362916c Make sure the first frame isn't dropped. 
If frames were delivered within the same millisecond as VideoCaptureImpl
was created, or the timestamp weren't granular enough then the first
frame would be mistakenly dropped because of having the same timestamp
as a previous one, even though there was no previous one.

BUG=
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5004 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 09:02:30 +00:00
kjellander@webrtc.org
eb61a851d5 Move audio_e2e_harness into include_tests==1 condition. 
To avoid compile errors when WebRTC is built as a part of
Chromium.

TEST=ran gclient runhooks locally.
BUG=none
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2609004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5003 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-21 08:40:56 +00:00
kjellander@webrtc.org
88a310886e Add audio_e2e_test target to tools.gyp 
The moving this GYP target out of webrtc.gyp in
https://code.google.com/p/webrtc/source/detail?r=4949
this should have been added into tools.gyp.

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

Review URL: https://webrtc-codereview.appspot.com/2539004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5002 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-19 18:10:04 +00:00
wu@webrtc.org
fb648da2b9 Protect _transportPtr, which can be accessed by different threads in the case of external transport. This change avoid the potential use-after-free, e.g. the case in the reported bug. 
BUG=2508
RISK=P1
TEST=try bots
R=henrika@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/2425004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5000 4adac7df-926f-26a2-2b94-8c16560cd09d
 2013-10-18 21:10:51 +00:00